A Signal is a FloatArray that represents a sampled function of time buffer. Signals support math operations.
Fill a Signal of the given size with a sum of sines at the given amplitudes and phases. The Signal will be normalized.
size 
the number of samples in the Signal. 
amplitudes 
an Array of amplitudes for each harmonic beginning with the fundamental. 
phases 
an Array of phases in radians for each harmonic beginning with the fundamental. 
Fill a Signal of the given size with a sum of Chebyshev polynomials at the given amplitudes. For eventual use in waveshaping by the Shaper ugen; see Shaper helpfile and Buffer:cheby too.
size 
the number of samples in the Signal. 
amplitudes 
an Array of amplitudes for each Chebyshev polynomial beginning with order 1. 
normalize 
a Boolean indicating whether to normalize the resulting Signal. If the zeroOffset argument is true, the normalization is done for use as a transfer function, using normalizeTransfer, otherwise it just uses normalize to make the absolute peak value 1. Default is true. 
zeroOffset 
a Boolean indicating whether to offset the middle of each polynomial to zero. If true, then a zero input will always result in a zero output when used as a waveshaper. If false, then the "raw" (unshifted) Chebyshev polynomials are used. Default is false. 
Fill a Signal of the given size with a Hanning window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
Fill a Signal of the given size with a Hamming window.
Signal.hammingWindow
had incorrect coefficients. To get the old behavior back, use Signal.hammingWindow_old
.size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
Fill a Signal of the given size with a Welch window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
Fill a Signal of the given size with a rectangular window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
Fourier Transform: Fill a Signal with the cosine table needed by the FFT methods. See also the instance methods fft and ifft.
This used to be Signal.hammingWindow
, but the coefficients were incorrect (making it a different window in the generalized Hamming window family). It is provided to assist in porting code to 3.11 and later.
For more information, check out the Guide: ExtraWindows
Fill a Signal of the given size with a Bartlett/triangular window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a BlackmanHarris window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a BlackmanNuttall window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a Gaussian window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
a 
the inverse of standard deviation (width) of the Gaussian. Higher means narrower; analogous to quality factor. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a Hanning window. Overwrote SCClassLibrary to have the symmetry argument.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a Hamming window. Overwrote SCClassLibrary to have the symmetry argument.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Synonymous with hanningWindow method with the extra symmetry argument.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a Kaiser window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
a 
the alpha value. Reduces to rectangular window at zero and narrows at higher values. Parameterized as described here: https://ccrma.stanford.edu/~jos/sasp/Kaiser_Window.html 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a Lanczos window.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fill a Signal of the given size with a Tukey window (taperedcosine window).
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
a 
the alpha value. At 0, reduces to Hann window. At 1, reduces to rectangular. Default to 0.5. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Fourier Transform: Fill a Signal with the cosine table needed by the RealFFT methods. See also the instance methods rfft and irfft.
Fourier Transform: Fill a Signal with the cosine table needed by the RealFFTTwo methods. See also the instance methods rfftTwo and irfftTwo.
Fill a Signal of the given size with a sum of cosines at the given amplitudes and phases. The Signal will be normalized.
size 
the number of samples in the Signal. 
amplitudes 
an Array of amplitudes for each harmonic beginning with the fundamental. 
phases 
an Array of phases in radians for each harmonic beginning with the fundamental. 
Fill a Signal of the given size with a sum of Hann enveloped cosines at the given amplitudes and phases. The Signal will be normalized.
size 
the number of samples in the Signal. 
amplitudes 
an Array of amplitudes for each harmonic beginning with the fundamental. 
phases 
an Array of phases in radians for each harmonic beginning with the fundamental. 
Fill a Signal of the given size with zeros. A synonym for *newClear.
size 
the number of samples in the Signal. 
Read a soundfile.
path 
a String representing the path of the soundfile to be read. 
size 
the number of samples. If size is 
startFrame 
the first frame of the soundfile to read. The default is 0, which is the beginning of the file. May be a negative number, which indicates the number of zeros to fill before reading the soundfile. 
channel 
soundfile channel to read. The default is 0, the first channel. 
Read a periodic waveform from a soundfile.
path 
a String representing the path of the soundfile to be read. 
size 
the number of samples. If size is 
frame 
center frame of the soundfile about which to read. I.e., the synthesis window is centered on this frame. The default is 0, which is the beginning of the file. 
channel 
soundfile channel to read. The default is 0, the first channel. 
freq 
analysis frequency of the waveform, in Hz. The default is 440 Hz. 
alpha 
Kaiser window alpha. The default is 3. 
winScale 
power of two scaling coefficient for the window size. The default is 1. 
freq
Successful synthesis of a periodic waveform from a soundfile depends largely on closely estimating the frequency of the waveform. The folding frequency of the Kaiser synthesis window is mapped to this analysis frequency.
alpha
Time domain smoothing is directly specified by the Kaiser window alpha. Choosing alpha = 0; winScale = 0
specifies a rectangular window, and allows us to view an unsmoothed period of the source waveform. Higher values of alpha mean more time domain smoothing.
winScale
The window scale argument scales the Kaiser window folding frequency with respect to the period of the waveform, like this:
winScale  waveform periods 
1  2 
0  1 
1  0.5 
The default, winScale = 1
, is optimimal in the sense that frequency domain artifacts are relected furthest from the harmonics of the periodic waveform of interest, without excessive oversampling.
Read a Hann enveloped wave packet from a soundfile.
path 
a String representing the path of the soundfile to be read. 
size 
the number of samples. If size is 
frame 
center frame of the soundfile about which to read. I.e., the synthesis window is centered on this frame. The default is 0, which is the beginning of the file. 
channel 
soundfile channel to read. The default is 0, the first channel. 
freq 
analysis frequency of the waveform, in Hz. The default is 440 Hz. 
alpha 
Kaiser window alpha. The default is 3. 
winScale 
power of two scaling coefficient for the window size. The default is 1. 
See *readWave discussion above.
Return a rectangular windowed sinc filter kernel.^{1}
size 
the number of samples in the Signal. 
freq 
the cutoff frequeny, in Hz. 
sampleRate 
sample rate, in Hz. 
Return a log shelf filter kernel.^{2} See also: FreqSpectrum: *logShelf.
size 
the number of samples in the Signal.  
freq0 
the frequeny, at which to begin transition from gainDC to gainNy, in Hz.  
freq1 
the frequeny, at which to end transition from gainDC to gainNy, in Hz.  
gainDC 
the gain at DC, in dB.  
gainNy 
the gain at Nyquist, in dB.  
phase 
phase response.
 
sampleRate 
sample rate, in Hz. 
Return an Array of gaussian windowed filter kernels.
size 
the number of samples in the Signals.  
freqs 
the crossover frequenies, in Hz.  
alpha 
window smoothing coefficient
 
sampleRate 
sample rate, in Hz. 
Return gaussian white noise, with a mean of 0 and standard deviation of 1.
See also SimpleNumber: gauss.
size 
the number of samples in the Signal. 
Return periodic powerlaw noise, aka fractional noise. Magnitude at Nyquist is normalized to 1.
size 
the number of samples in the Signal.  
beta 
power spectral density.

Return a rectangular windowed sinus powerlaw chirp.^{3}
size 
the number of samples in the Signal.  
freq0 
initial frequeny, in Hz.  
freq1 
final frequeny, in Hz.  
phase 
initial phase, in radians.  
beta 
power spectral density.
 
sampleRate 
sample rate, in Hz. 
Synthesized in the time domain.
Return a rectangular windowed sinus powerlaw chirp.
size 
the number of samples in the Signal.  
freq0 
initial frequeny, in Hz.  
freq1 
final frequeny, in Hz.  
phase 
initial phase, in radians.  
beta 
power spectral density.
 
sampleRate 
sample rate, in Hz. 
Synthesized in the time domain.
Return a rectangular windowed complex analytic sinus powerlaw chirp.
size 
the number of samples in the Signal.  
freq0 
initial frequeny, in Hz.  
freq1 
final frequeny, in Hz.  
phase 
initial phase, in radians.  
beta 
power spectral density.
 
sampleRate 
sample rate, in Hz. 
Synthesized in the time domain.
Return a complex analytic Signal of the given size with a sum of cosines and a sum of sines at the given amplitudes and phases. The Signal will be normalized.
size 
the number of samples in the Signal. 
amplitudes 
an Array of amplitudes for each harmonic beginning with the fundamental. 
phases 
an Array of phases in radians for each harmonic beginning with the fundamental. 
Hilbert Transform: Return complex Hilbert Transform coefficients.
size 
the number of samples in the Signal. 
pad 
the number of samples of the size that is zero padding. 
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Return an Array of Higher Order Ambisonic signal (HOA) nearfield effect (NFE), distance radial filters collected by Associated Legendre degree (ℓ).
size 
The number of samples in the Signals. 
radius 
Radius, in meters. 
order 
Ambisonic order. 
sampleRate 
Sample rate, in Hz. 
speedOfSound 
Speed of sound, in meters per second. 
Offers FIR coefficients, equivalent to the IIR coefficients returned by NFECoeffs: dist. Implemented as a frequency sampling design, with coefficients returned by HoaOrder: distWeights. See also: FreqSpectrum: *hoaDist.
Return an Array of Higher Order Ambisonic signal (HOA) nearfield effect (NFE), control radial filters collected by Associated Legendre degree (ℓ).
size 
The number of samples in the Signals. 
encRadius 
Encoding radius, in meters. 
decRadius 
Decoding radius, in meters. 
order 
Ambisonic order. 
sampleRate 
Sample rate, in Hz. 
speedOfSound 
Speed of sound, in meters per second. 
Offers FIR coefficients, equivalent to the IIR coefficients returned by NFECoeffs: ctrl. Implemented as a frequency sampling design, with coefficients returned by HoaOrder: ctrlWeights. See also: FreqSpectrum: *hoaCtrl.
Return an Array of Higher Order Ambisonic signal (HOA) nearfield effect (NFE), focalisation radial filters collected by Associated Legendre degree (ℓ).
size 
The number of samples in the Signals.  
radius 
Radius, in meters.  
order 
Ambisonic order.  
window 
Angular weighting window.
 
sampleRate 
Sample rate, in Hz.  
speedOfSound 
Speed of sound, in meters per second. 
Offers FIR coefficients; implemented as a linear phase frequency sampling design, with coefficients returned by HoaOrder: foclWeights. See also: FreqSpectrum: *hoaFocl.
Return an Array of Higher Order Ambisonic signal (HOA) filters combining multiband beamforming and nearfield effect (NFE) focalisation, collected by Associated Legendre degree (ℓ).
size 
The number of samples in the Signals.  
radius 
Radius, in meters. Set to  
beamDict 
A dictionary specifying beam shapes and edge frequencies. See Beaming & Decoder Matching and discussion below.  
dim 
The number of dimensions: 2D or 3D.  
match 
Matching criteria, see Beaming & Decoder Matching:
NOTE: Prepend f to include focalisation in normalisation. E.g., \frms .  
numChans 
Number of loudspeakers. NOTE: Must be set when choosing match: \energy .  
order 
Ambisonic order.  
window 
Focalisation angular weighting window. See *hoaFocl. NOTE: Ignored if radius is set to nil .  
sampleRate 
Sample rate, in Hz.  
speedOfSound 
Speed of sound, in meters per second. 
Offers FIR coefficients; implemented as a linear phase frequency sampling design, with coefficients returned by *logShelf and *hoaFocl. See also: FreqSpectrum: *hoaMultiBandFocl for design examples.
Plot the Signal in a window. The arguments are not required and if not given defaults will be used.
For details, see plot
Loads the signal into a buffer on the server and plays it. Returns the buffer so you can free it again.
loop 
A Boolean whether to loop the entire signal or play it once. Default is false. 
mul 
volume at which to play it, 0.2 by default. 
numChannels 
if the signal is an interleaved multichannel file, number of channels, default is 1. 
server 
the server on which to load the signal into a buffer. 
Fill the Signal with a function evaluated over an interval.
function 
a function that should calculate the value of a sample. The function is called with three arguments:
As arguments, three values are passed to the function: the current input value (abscissa), the old value (if the signal is overwritten), and the index. 
start 
the starting value of the interval. 
end 
the ending value of the interval. 
Convert the Signal into a Wavetable.
Fill the Signal with a value.
Scale the Signal by a factor in place.
Offset the Signal by a value in place.
Return the peak absolute value of a Signal.
Normalize the Signal in place such that the maximum absolute peak value is 1.
Normalizes a transfer function so that the center value of the table is offset to zero and the absolute peak value is 1. Transfer functions are meant to be used in the Shaper ugen.
Invert the Signal in place.
Reverse a subrange of the Signal in place.
Fade a subrange of the Signal in place.
Return the integral of a signal.
Add a signal to myself starting at the index. If the other signal is too long only the first part is overdubbed.
Write a signal to myself starting at the index. If the other signal is too long only the first part is overdubbed.
Blend two signals by some proportion.
Perform an FFT on a real and imaginary signal in place. See also the class method *fftCosTable.
Perform an inverse FFT on a real and imaginary signal in place. See also the class method *fftCosTable.
Return a complex RealFFT spectrum from a complex FFT spectrum. See also the instance method rfftToFft.
Perform a RealFFT on a real signal.^{8} See also the class method *rfftCosTable.
Perform an inverse RealFFT on a real and imaginary signal. See also the class method *rfftCosTable.
Return a complex FFT spectrum from a complex RealFFT spectrum. See also the instance method fftToRfft.
Perform a RealFFT on two real signals. See also the class method *rfftTwoCosTable.
Perform an inverse RealFFT on two real and imaginary signals. See also the class method *rfftTwoCosTable.
Perform a DFT on a real and imaginary signal.
size
is not restricted to a power of two.imag 
imaginary signal.  
method 

Perform an inverse DFT on a real and imaginary signal.
size
is not restricted to a power of two.imag 
imaginary signal.  
method 

Perform a Zoom DFT on a real and imaginary signal.
Signal.size
is not restricted to a power of two.imag 
imaginary signal. 
zoomsize 
the number of coefficients to return. 
k0 
lowest DFT coefficient. Not required to be an integer. 
k1 
highest DFT coefficient. Not required to be an integer. 
Perform a Zoom DFT on a real signal.
Signal.size
is not restricted to a power of two.zoomsize 
the number of coefficients to return. 
k0 
lowest DFT coefficient. Not required to be an integer. 
k1 
highest DFT coefficient. Not required to be an integer. 
Return individual terms of a DFT on a real and imaginary signal.^{9}
Signal.size
is not restricted to a power of two.imag 
imaginary signal.  
k 
DFT coefficient(s) as a single value or an Array. Not required to be an integer.  
method 

Return individual terms of a DFT on a real signal.
Signal.size
is not restricted to a power of two.k 
DFT coefficient(s) as a single value or an Array. Not required to be an integer.  
method 

Signal will respond to unary operators by returning a new Signal.
The flip operator.^{10}
Signal will respond to binary operators by returning a new Signal.
Thresholding replaces every value < threshold with 0.
Bilateral thresholding.
thresh 
When the 
adverb 
Optional, for processing Collections. See Adverbs for Binary Operators. 
Add a single cosine to myself with the given harmonic, amplitude and phase.
A sum of cosines with the given amplitudes and phases, returned in place.
amplitudes 
an Array of amplitudes for each harmonic beginning with the fundamental. 
phases 
an Array of phases in radians for each harmonic beginning with the fundamental. 
A sum of cosines with the given harmonics, amplitudes and phases, returned in place.
list 
a rank 2 collection containing harmonic numbers, amplitudes and phases. 
A sum of Hann enveloped cosines at the given amplitudes and phases, returned in place.
amplitudes 
an Array of amplitudes for each harmonic beginning with the fundamental. 
phases 
an Array of phases in radians for each harmonic beginning with the fundamental. 
A sum of Hann enveloped cosines with the given harmonics, amplitudes and phases, returned in place.
list 
a rank 2 collection containing harmonic numbers, amplitudes and phases. 
Given a periodic waveform, returns a Hann enveloped wave packet. The receiver is unchanged.
Given a superposition density window, returns a constantoverlapadd (COLA) window.^{11} The receiver is unchanged.
Returns a new Signal whose elements are repeated sequences of the receiver, up to size length. The receiver is unchanged.
Returns a new Signal resampled in the frequency domain to a new size. The receiver is unchanged.
Remove the DC component the Signal in place.
Writes the entire Signal to a file, where every chunk of four bytes is a 32bit floating point sample.
Writes the entire Signal to a soundfile.
path 
a String representing the path of the soundfile to be written. 
headerFormat 
a String for the soundfile format. Valid strings are listed at SoundFile: headerFormat. If not given, the default 
sampleFormat 
a String for the sample format. Valid strings are listed at SoundFile: sampleFormat. If not given, the default 
sampleRate 
an Integer sample rate (44100 by default). 
Convolve myself with a signal.
aSignal 
the multiplier.  
method 

Periodic convolve myself with a signal.
aSignal 
the multiplier.  
method 

Partition convolve myself with a signal.
aSignal 
the multiplier.  
size 
the partition size.  
method 

Hilbert Transform: Return a complex analytic signal from a real signal.^{12}
Return the instantaneous frequency from a real signal.^{13}
mindb 
a minimum value in dB to clip amplitude response to. Prevents singularities. 
sampleRate 
sample rate, in Hz. 
Return a complex signal with the given real and imaginary parts.
Rotate the Signal by a value in radians, in place.
Rotate the phase of Signal by a value in radians, in place.
Return a linear phase kernel, preserving the magnitude, in place.
sym 
a Boolean indicating whether the window is symmetric or periodic. Default is false. 
Return a minimum phase kernel, preserving the magnitude, in place.^{14}
mindb 
a minimum value in dB to clip amplitude response to. Reduces time aliasing. 
oversample 
time oversampling scale factor. May be a float or an integer. Reduces time aliasing. 
Return a gaussian noise phase kernel, preserving the magnitude, in place.
Return the even part of a signal.^{15}
Return the odd part of a signal.
Return the RMS of a signal.
Return the peak of the frequency domain magnitude of a signal.
oversample 
time oversampling scale factor. May be a float or an integer. 
Normalize the Signal in place such that the maximum magnitude in the frequency domain is 1.
oversample 
time oversampling scale factor. May be a float or an integer. 
Return the real part of the cepstrum of a real signal.^{16}
Return the real part of the inverse of the real part of the cepstrum.
Perform a Chirp zTransform on a real and imaginary signal.^{17}
imag 
imaginary signal. 
cwtsize 
the number of coefficients to return. Defaults to 
step 
ratio between points along the zplane spiral contour of interest. Defaults to 
start 
complex starting point on that contour. Defaults to NOTE: The default nil values of cwtsize , step , and start return the DFT. 
Perform a Chirp zTransform on a real signal.
cwtsize 
the number of coefficients to return. Defaults to 
step 
ratio between points along the zplane spiral contour of interest. Defaults to 
start 
complex starting point on that contour. Defaults to NOTE: The default nil values of cwtsize , step , and start return the DFT. 