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I hope that someday Octave will include more signal processing functions. If you would like to help improve Octave in this area, please contact [email protected].
If x is a vector,
detrend (
x,
p)
removes the best fit of a polynomial of order p from the data x.If x is a matrix,
detrend (
x,
p)
does the same for each column in x.The second argument is optional. If it is not specified, a value of 1 is assumed. This corresponds to removing a linear trend.
Compute the FFT of a using subroutines from Fftw. The FFT is calculated along the first non-singleton dimension of the array. Thus if a is a matrix,
fft (
a)
computes the FFT for each column of a.If called with two arguments, n is expected to be an integer specifying the number of elements of a to use, or an empty matrix to specify that its value should be ignored. If n is larger than the dimension along which the FFT is calculated, then a is resized and padded with zeros. Otherwise, ifn is smaller than the dimension along which the FFT is calculated, then a is truncated.
If called with three agruments, dim is an integer specifying the dimension of the matrix along which the FFT is performed
Compute the inverse FFT of a using subroutines from Fftw. The inverse FFT is calculated along the first non-singleton dimension of the array. Thus if a is a matrix,
fft (
a)
computes the inverse FFT for each column of a.If called with two arguments, n is expected to be an integer specifying the number of elements of a to use, or an empty matrix to specify that its value should be ignored. If n is larger than the dimension along which the inverse FFT is calculated, then a is resized and padded with zeros. Otherwise, ifn is smaller than the dimension along which the inverse FFT is calculated, then a is truncated.
If called with three agruments, dim is an integer specifying the dimension of the matrix along which the inverse FFT is performed
Compute the two dimensional FFT of a using subroutines from Fftw. The optional arguments n and m may be used specify the number of rows and columns of a to use. If either of these is larger than the size of a, a is resized and padded with zeros.
If a is a multi-dimensional matrix, each two-dimensional sub-matrix of a is treated seperately
Compute the inverse two dimensional FFT of a using subroutines from Fftw. The optional arguments n and m may be used specify the number of rows and columns of a to use. If either of these is larger than the size of a, a is resized and padded with zeros.
If a is a multi-dimensional matrix, each two-dimensional sub-matrix of a is treated seperately
Compute the N dimensional FFT of a using subroutines from Fftw. The optional vector argument size may be used specify the dimensions of the array to be used. If an element of size is smaller than the corresponding dimension, then the dimension is truncated prior to performing the FFT. Otherwise if an element of size is larger than the corresponding dimension a is resized and padded with zeros.
Compute the invesre N dimensional FFT of a using subroutines from Fftw. The optional vector argument size may be used specify the dimensions of the array to be used. If an element of size is smaller than the corresponding dimension, then the dimension is truncated prior to performing the inverse FFT. Otherwise if an element of size is larger than the corresponding dimension a is resized and padded with zeros.
Return the convolution of the vectors a and b, as a vector with length equal to the
length (a) + length (b) - 1
. If a and b are the coefficient vectors of two polynomials, the returned value is the coefficient vector of the product polynomial.The computation uses the FFT by calling the function
fftfilt
. If the optional argument n is specified, an N-point FFT is used.
With two arguments,
fftfilt
filters x with the FIR filter b using the FFT.Given the optional third argument, n,
fftfilt
uses the overlap-add method to filter x with b using an N-point FFT.If x is a matrix, filter each column of the matrix.
Return the solution to the following linear, time-invariant difference equation:
N M SUM a(k+1) y(n-k) = SUM b(k+1) x(n-k) for 1<=n<=length(x) k=0 k=0where N=length(a)-1 and M=length(b)-1. over the first non-singleton dimension of x or over dim if supplied. An equivalent form of this equation is:
N M y(n) = - SUM c(k+1) y(n-k) + SUM d(k+1) x(n-k) for 1<=n<=length(x) k=1 k=0where c = a/a(1) and d = b/a(1).
If the fourth argument si is provided, it is taken as the initial state of the system and the final state is returned as sf. The state vector is a column vector whose length is equal to the length of the longest coefficient vector minus one. If si is not supplied, the initial state vector is set to all zeros.
In terms of the z-transform, y is the result of passing the discrete- time signal x through a system characterized by the following rational system function:
M SUM d(k+1) z^(-k) k=0 H(z) = ---------------------- N 1 + SUM c(k+1) z(-k) k=1
Return the complex frequency response h of the rational IIR filter whose numerator and denominator coefficients are b and a, respectively. The response is evaluated at n angular frequencies between 0 and 2*pi.
The output value w is a vector of the frequencies.
If the fourth argument is omitted, the response is evaluated at frequencies between 0 and pi.
If n is omitted, a value of 512 is assumed.
If a is omitted, the denominator is assumed to be 1 (this corresponds to a simple FIR filter).
For fastest computation, n should factor into a small number of small primes. — Function File: h = freqz (b, a, w)
Evaluate the response at the specific frequencies in the vector w. The values for w are measured in radians. — Function File: [...] = freqz (..., Fs)
Return frequencies in Hz instead of radians assuming a sampling rate Fs. If you are evaluating the response at specific frequencies w, those frequencies should be requested in Hz rather than radians. — Function File: freqz (...)
Plot the pass band, stop band and phase response of h rather than returning them.
Unwrap radian phases by adding multiples of 2*pi as appropriate to remove jumps greater than tol. tol defaults to pi.
Unwrap will unwrap along the first non-singleton dimension of a, unless the optional argument dim is given, in which case the data will be unwrapped along this dimension
Fit an ARCH regression model to the time series y using the scoring algorithm in Engle's original ARCH paper. The model is
y(t) = b(1) * x(t,1) + ... + b(k) * x(t,k) + e(t), h(t) = a(1) + a(2) * e(t-1)^2 + ... + a(p+1) * e(t-p)^2in which e(t) is N(0, h(t)), given a time-series vector y up to time t-1 and a matrix of (ordinary) regressors x up to t. The order of the regression of the residual variance is specified by p.
If invoked as
arch_fit (
y,
k,
p)
with a positive integer k, fit an ARCH(k, p) process, i.e., do the above with the t-th row of x given by[1, y(t-1), ..., y(t-k)]Optionally, one can specify the number of iterations iter, the updating factor gamma, and initial values a0 and b0 for the scoring algorithm.
Simulate an ARCH sequence of length t with AR coefficients b and CH coefficients a. I.e., the result y(t) follows the model
y(t) = b(1) + b(2) * y(t-1) + ... + b(lb) * y(t-lb+1) + e(t),where e(t), given y up to time t-1, is N(0, h(t)), with
h(t) = a(1) + a(2) * e(t-1)^2 + ... + a(la) * e(t-la+1)^2
For a linear regression model
y = x * b + eperform a Lagrange Multiplier (LM) test of the null hypothesis of no conditional heteroscedascity against the alternative of CH(p).
I.e., the model is
y(t) = b(1) * x(t,1) + ... + b(k) * x(t,k) + e(t),given y up to t-1 and x up to t, e(t) is N(0, h(t)) with
h(t) = v + a(1) * e(t-1)^2 + ... + a(p) * e(t-p)^2,and the null is a(1) == ... == a(p) == 0.
If the second argument is a scalar integer, k, perform the same test in a linear autoregression model of order k, i.e., with
[1, y(t-1), ..., y(t-k)]as the t-th row of x.
Under the null, LM approximately has a chisquare distribution with p degrees of freedom and pval is the p-value (1 minus the CDF of this distribution at LM) of the test.
If no output argument is given, the p-value is displayed.
Return a simulation of the ARMA model
x(n) = a(1) * x(n-1) + ... + a(k) * x(n-k) + e(n) + b(1) * e(n-1) + ... + b(l) * e(n-l)in which k is the length of vector a, l is the length of vector b and e is gaussian white noise with variance v. The function returns a vector of length t.
The optional parameter n gives the number of dummy x(i) used for initialization, i.e., a sequence of length t+n is generated and x(n+1:t+n) is returned. If n is omitted, n = 100 is used.
Return the autocorrelations from lag 0 to h of vector x. If h is omitted, all autocorrelations are computed. If x is a matrix, the autocorrelations of each column are computed.
Return the autocovariances from lag 0 to h of vector x. If h is omitted, all autocovariances are computed. If x is a matrix, the autocovariances of each column are computed.
Given a time series (vector) y, return a matrix with ones in the first column and the first k lagged values of y in the other columns. I.e., for t > k,
[1,
y(
t-1), ...,
y(
t-
k)]
is the t-th row of the result. The resulting matrix may be used as a regressor matrix in autoregressions.
Return the filter coefficients of a Bartlett (triangular) window of length m.
For a definition of the Bartlett window, see e.g. A. V. Oppenheim & R. W. Schafer, "Discrete-Time Signal Processing".
Return the filter coefficients of a Blackman window of length m.
For a definition of the Blackman window, see e.g. A. V. Oppenheim & R. W. Schafer, "Discrete-Time Signal Processing".
Return the estimator d for the differencing parameter of an integrated time series.
The frequencies from [2*pi*a/t, 2*pi*b/T] are used for the estimation. If b is omitted, the interval [2*pi/T, 2*pi*a/T] is used. If both b and a are omitted then a = 0.5 * sqrt (T) and b = 1.5 * sqrt (T) is used, where T is the sample size. If x is a matrix, the differencing parameter of each column is estimated.
The estimators for all frequencies in the intervals described above is returned in dd. The value of d is simply the mean of dd.
Reference: Brockwell, Peter J. & Davis, Richard A. Time Series: Theory and Methods Springer 1987.
Perform one step of the Durbin-Levinson algorithm.
The vector c specifies the autocovariances
[gamma_0, ..., gamma_t]
from lag 0 to t, oldphi specifies the coefficients based on c(t-1) and oldv specifies the corresponding error.If oldphi and oldv are omitted, all steps from 1 to t of the algorithm are performed.
Perform a shift of the vector v, for use with the
fft
andifft
functions, in order the move the frequency 0 to the center of the vector or matrix.If v is a vector of N elements corresponding to N time samples spaced of Dt each, then
fftshift (fft (
v))
corresponds to frequenciesf = ((1:N) - ceil(N/2)) / N / DtIf v is a matrix, the same holds for rows and columns. If v is an array, then the same holds along each dimension.
The optional dim argument can be used to limit the dimension along which the permutation occurs.
Compute the fractional differences (1-L)^d x where L denotes the lag-operator and d is greater than -1.
Return the filter coefficients of a Hamming window of length m.
For a definition of the Hamming window, see e.g. A. V. Oppenheim & R. W. Schafer, "Discrete-Time Signal Processing".
Return the filter coefficients of a Hanning window of length m.
For a definition of this window type, see e.g. A. V. Oppenheim & R. W. Schafer, "Discrete-Time Signal Processing".
Estimate the Hurst parameter of sample x via the rescaled range statistic. If x is a matrix, the parameter is estimated for every single column.
For a data matrix x from a sample of size n, return the periodogram.
Rectangular lag window. Subfunction used for spectral density estimation.
Rectangular spectral window. Subfunction used for spectral density estimation.
Return a sinetone of frequency freq with length of sec seconds at sampling rate rate and with amplitude ampl. The arguments freq and ampl may be vectors of common size.
Defaults are rate = 8000, sec = 1 and ampl = 64.
Return an m-element vector with i-th element given by
sin (2 * pi * (
i+
d-1) /
n)
.The default value for d is 0 and the default value for n is m.
Return the spectral density estimator given a vector of autocovariances c, window name win, and bandwidth, b.
The window name, e.g.,
"triangle"
or"rectangle"
is used to search for a function called win_sw
.If win is omitted, the triangle window is used. If b is omitted,
1 / sqrt (length (
x))
is used.
Return the spectral density estimator given a data vector x, window name win, and bandwidth, b.
The window name, e.g.,
"triangle"
or"rectangle"
is used to search for a function called win_sw
.If win is omitted, the triangle window is used. If b is omitted,
1 / sqrt (length (
x))
is used.
Compute the short-term Fourier transform of the vector x with num_coef coefficients by applying a window of win_size data points and an increment of inc points.
Before computing the Fourier transform, one of the following windows is applied:
- hanning
- w_type = 1
- hamming
- w_type = 2
- rectangle
- w_type = 3
The window names can be passed as strings or by the w_type number.
If not all arguments are specified, the following defaults are used: win_size = 80, inc = 24, num_coef = 64, and w_type = 1.
y
= stft (
x, ...)
returns the absolute values of the Fourier coefficients according to the num_coef positive frequencies.
[
y,
c] = stft (x, ...)
returns the entire STFT-matrix y and a 3-element vector c containing the window size, increment, and window type, which is needed by the synthesis function.
Compute a signal from its short-time Fourier transform y and a 3-element vector c specifying window size, increment, and window type.
The values y and c can be derived by
[y, c] = stft (x , ...)
Triangular lag window. Subfunction used for spectral density estimation.