octave-signal 1.3.2-5 / usr / share / octave / packages / signal-1.3.2 / doc-cache

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ar_psd


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 -- Function File: ar_psd (A, V)
 -- Function File: ar_psd (A, V, FREQ)
 -- Function File: ar_psd (A, V, FREQ, FS)
 -- Function File: ar_psd (..., RANGE)
 -- Function File: ar_psd (..., METHOD)
 -- Function File: ar_psd (..., PLOTTYPE)
 -- Function File: [PSD, F_OUT] = ar_psd (...)

     Calculate the power spectrum of the autoregressive model

                                 M
          x(n) = sqrt(v).e(n) + SUM a(k).x(n-k)
                                k=1

     where x(n) is the output of the model and e(n) is white noise.
     This function is intended for use with '[a, v, k] = arburg (x,
     poles, criterion)' which use the Burg (1968) method to calculate a
     "maximum entropy" autoregressive model of X.

     If the FREQ argument is a vector (of frequencies) the spectrum is
     calculated using the polynomial method and the METHOD argument is
     ignored.  For scalar FREQ, an integer power of 2, or METHOD =
     "FFT", causes the spectrum to be calculated by FFT. Otherwise, the
     spectrum is calculated as a polynomial.  It may be computationally
     more efficient to use the FFT method if length of the model is not
     much smaller than the number of frequency values.  The spectrum is
     scaled so that spectral energy (area under spectrum) is the same as
     the time-domain energy (mean square of the signal).

     ARGUMENTS: All but the first two arguments are optional and may be
     empty.
        * A list of M=(order+1) autoregressive model coefficients.  The
          first element of "ar_coeffs" is the zero-lag coefficient,
          which always has a value of 1.
        * V square of the moving-average coefficient of the AR model.
        * FREQ frequencies at which power spectral density is
          calculated, or a scalar indicating the number of uniformly
          distributed frequency values at which spectral density is
          calculated.  (default = 256)
        * FS sampling frequency (Hertz) (default=1)

     CONTROL-STRING ARGUMENTS - each of these arguments is a character
     string.  Control-string arguments can be in any order after the
     other arguments.

     Range:

     'half', 'onesided' : frequency range of the spectrum is from zero
     up to but not including sample_f/2.  Power from negative
     frequencies is added to the positive side of the spectrum.
     'whole', 'twosided' : frequency range of the spectrum is
     -sample_f/2 to sample_f/2, with negative frequencies stored in
     "wrap around" order after the positive frequencies; e.g.
     frequencies for a 10-point 'twosided' spectrum are 0 0.1 0.2 0.3
     0.4 0.5 -0.4 -0.3 -0.2 -0.1 'shift', 'centerdc' : same as 'whole'
     but with the first half of the spectrum swapped with second half to
     put the zero-frequency value in the middle.  (See "help fftshift".
     If "freq" is vector, 'shift' is ignored.  If model coefficients
     "ar_coeffs" are real, the default range is 'half', otherwise
     default range is 'whole'.

     Method:

     'fft': use FFT to calculate power spectrum.  'poly': calculate
     power spectrum as a polynomial of 1/z N.B. this argument is ignored
     if the "freq" argument is a vector.  The default is 'poly' unless
     the "freq" argument is an integer power of 2.

     Plot type:

     'plot', 'semilogx', 'semilogy', 'loglog', 'squared' or 'db':
     specifies the type of plot.  The default is 'plot', which means
     linear-linear axes.  'squared' is the same as 'plot'.  'dB' plots
     "10*log10(psd)".  This argument is ignored and a spectrum is not
     plotted if the caller requires a returned value.

     RETURNED VALUES: If returned values are not required by the caller,
     the spectrum is plotted and nothing is returned.
        * PSD estimate of power-spectral density
        * F_OUT frequency values

     REFERENCE [1] Equation 2.28 from Steven M. Kay and Stanley Lawrence
     Marple Jr.: "Spectrum analysis - a modern perspective", Proceedings
     of the IEEE, Vol 69, pp 1380-1419, Nov., 1981


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Calculate the power spectrum of the autoregressive model



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arburg


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 -- Function File: [A, V, K] = arburg (X, POLES)
 -- Function File: [A, V, K] = arburg (X, POLES, CRITERION)

     Calculate coefficients of an autoregressive (AR) model of complex
     data X using the whitening lattice-filter method of Burg (1968).
     The inverse of the model is a moving-average filter which reduces X
     to white noise.  The power spectrum of the AR model is an estimate
     of the maximum entropy power spectrum of the data.  The function
     'ar_psd' calculates the power spectrum of the AR model.

     ARGUMENTS:
        * X sampled data
        * POLES number of poles in the AR model or limit to the number
          of poles if a valid CRITERION is provided.
        * CRITERION model-selection criterion.  Limits the number of
          poles so that spurious poles are not added when the whitened
          data has no more information in it (see Kay & Marple, 1981).
          Recognized values are 'AKICc' - approximate corrected Kullback
          information criterion (recommended), 'KIC' - Kullback
          information criterion 'AICc' - corrected Akaike information
          criterion 'AIC' - Akaike information criterion 'FPE' - final
          prediction error" criterion The default is to NOT use a
          model-selection criterion

     RETURNED VALUES:
        * A list of (P+1) autoregression coefficients; for data input
          x(n) and white noise e(n), the model is

                                     P+1
               x(n) = sqrt(v).e(n) + SUM a(k).x(n-k)
                                     k=1

          V mean square of residual noise from the whitening operation
          of the Burg lattice filter.
        * K reflection coefficients defining the lattice-filter
          embodiment of the model

     HINTS:

     (1) arburg does not remove the mean from the data.  You should
     remove the mean from the data if you want a power spectrum.  A
     non-zero mean can produce large errors in a power-spectrum
     estimate.  See "help detrend".  (2) If you don't know what the
     value of "poles" should be, choose the largest (reasonable) value
     you could want and use the recommended value, criterion='AKICc', so
     that arburg can find it.  E.g.  arburg(x,64,'AKICc') The AKICc has
     the least bias and best resolution of the available model-selection
     criteria.  (3) Autoregressive and moving-average filters are stored
     as polynomials which, in matlab, are row vectors.

     NOTE ON SELECTION CRITERION:

     AIC, AICc, KIC and AKICc are based on information theory.  They
     attempt to balance the complexity (or length) of the model against
     how well the model fits the data.  AIC and KIC are biased estimates
     of the asymmetric and the symmetric Kullback-Leibler divergence
     respectively.  AICc and AKICc attempt to correct the bias.  See
     reference [4].

     REFERENCES:

     [1] John Parker Burg (1968) "A new analysis technique for time
     series data", NATO advanced study Institute on Signal Processing
     with Emphasis on Underwater Acoustics, Enschede, Netherlands, Aug.
     12-23, 1968.

     [2] Steven M. Kay and Stanley Lawrence Marple Jr.: "Spectrum
     analysis - a modern perspective", Proceedings of the IEEE, Vol 69,
     pp 1380-1419, Nov., 1981

     [3] William H. Press and Saul A. Teukolsky and William T.
     Vetterling and Brian P. Flannery "Numerical recipes in C, The art
     of scientific computing", 2nd edition, Cambridge University Press,
     2002 -- Section 13.7.

     [4] Abd-Krim Seghouane and Maiza Bekara "A small sample model
     selection criterion based on Kullback's symmetric divergence", IEEE
     Transactions on Signal Processing, Vol.  52(12), pp 3314-3323, Dec.
     2004

     See also: ar_psd.


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Calculate coefficients of an autoregressive (AR) model of complex data X
using t



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aryule


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 -- Function File: A = aryule (X, P)
 -- Function File: [A, V, K] = aryule (X, P)
     Fit an AR (P)-model with Yule-Walker estimates.
     X
          data vector to estimate
     A
          AR coefficients
     V
          variance of white noise
     K
          reflection coefficients for use in lattice filter

     The power spectrum of the resulting filter can be plotted with
     pyulear(x, p), or you can plot it directly with ar_psd(a,v,...).

     See also: pyulear, power, freqz, impz - for observing
     characteristics of the model arburg - for alternative spectral
     estimators

     Example: Use example from arburg, but substitute aryule for arburg.

     Note: Orphanidis '85 claims lattice filters are more tolerant of
     truncation errors, which is why you might want to use them.
     However, lacking a lattice filter processor, I haven't tested that
     the lattice filter coefficients are reasonable.


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Fit an AR (P)-model with Yule-Walker estimates.



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barthannwin


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 -- Function File: barthannwin (M)
     Return the filter coefficients of a modified Bartlett-Hann window
     of length M.

     See also: rectwin, bartlett.


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Return the filter coefficients of a modified Bartlett-Hann window of
length M.



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besselap


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 -- Function File: [ZERO, POLE, GAIN] = besselap (N)
     Return bessel analog filter prototype.

     References:

     http://en.wikipedia.org/wiki/Bessel_polynomials


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Return bessel analog filter prototype.



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besself


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 -- Function File: [B, A] = besself (N, W)
 -- Function File: [B, A] = besself (N, W, "high")
 -- Function File: [Z, P, G] = besself (...)
 -- Function File: [A, B, C, D] = besself (...)
 -- Function File: [...] = besself (..., "z")
     Generate a Bessel filter.  Default is a Laplace space (s) filter.

     [b,a] = besself(n, Wc) low pass filter with cutoff pi*Wc radians

     [b,a] = besself(n, Wc, 'high') high pass filter with cutoff pi*Wc
     radians

     [z,p,g] = besself(...)  return filter as zero-pole-gain rather than
     coefficients of the numerator and denominator polynomials.

     [...]  = besself(...,'z') return a discrete space (Z) filter, W
     must be less than 1.

     [a,b,c,d] = besself(...)  return state-space matrices

     References:

     Proakis & Manolakis (1992).  Digital Signal Processing.  New York:
     Macmillan Publishing Company.


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Generate a Bessel filter.



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bilinear


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 -- Function File: [ZB, ZA] = bilinear (SB, SA, T)
 -- Function File: [ZB, ZA] = bilinear (SZ, SP, SG, T)
 -- Function File: [ZZ, ZP, ZG] = bilinear (...)
     Transform a s-plane filter specification into a z-plane
     specification.  Filters can be specified in either zero-pole-gain
     or transfer function form.  The input form does not have to match
     the output form.  1/T is the sampling frequency represented in the
     z plane.

     Note: this differs from the bilinear function in the signal
     processing toolbox, which uses 1/T rather than T.

     Theory: Given a piecewise flat filter design, you can transform it
     from the s-plane to the z-plane while maintaining the band edges by
     means of the bilinear transform.  This maps the left hand side of
     the s-plane into the interior of the unit circle.  The mapping is
     highly non-linear, so you must design your filter with band edges
     in the s-plane positioned at 2/T tan(w*T/2) so that they will be
     positioned at w after the bilinear transform is complete.

     The following table summarizes the transformation:

          +---------------+-----------------------+----------------------+
          | Transform     | Zero at x             | Pole at x            |
          |    H(S)       |   H(S) = S-x          |    H(S)=1/(S-x)      |
          +---------------+-----------------------+----------------------+
          |       2 z-1   | zero: (2+xT)/(2-xT)   | zero: -1             |
          |  S -> - ---   | pole: -1              | pole: (2+xT)/(2-xT)  |
          |       T z+1   | gain: (2-xT)/T        | gain: (2-xT)/T       |
          +---------------+-----------------------+----------------------+

     With tedious algebra, you can derive the above formulae yourself by
     substituting the transform for S into H(S)=S-x for a zero at x or
     H(S)=1/(S-x) for a pole at x, and converting the result into the
     form:

             H(Z)=g prod(Z-Xi)/prod(Z-Xj)

     Please note that a pole and a zero at the same place exactly
     cancel.  This is significant since the bilinear transform creates
     numerous extra poles and zeros, most of which cancel.  Those which
     do not cancel have a "fill-in" effect, extending the shorter of the
     sets to have the same number of as the longer of the sets of poles
     and zeros (or at least split the difference in the case of the band
     pass filter).  There may be other opportunistic cancellations but I
     will not check for them.

     Also note that any pole on the unit circle or beyond will result in
     an unstable filter.  Because of cancellation, this will only happen
     if the number of poles is smaller than the number of zeros.  The
     analytic design methods all yield more poles than zeros, so this
     will not be a problem.

     References:

     Proakis & Manolakis (1992).  Digital Signal Processing.  New York:
     Macmillan Publishing Company.


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Transform a s-plane filter specification into a z-plane specification.



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bitrevorder


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 -- Function File: Y = bitrevorder (X)
 -- Function File: [Y I] = bitrevorder (X)
     Reorder the elements of the vector X in bit-reversed order.
     Equivalent to calling 'digitrevorder (X, 2)'.

     See also: digitrevorder, fft, ifft.


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Reorder the elements of the vector X in bit-reversed order.



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blackmanharris


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 -- Function File: blackmanharris (M)
 -- Function File: blackmanharris (M, "periodic")
 -- Function File: blackmanharris (M, "symmetric")
     Return the filter coefficients of a Blackman-Harris window of
     length M.

     If the optional argument '"periodic"' is given, the periodic form
     of the window is returned.  This is equivalent to the window of
     length M+1 with the last coefficient removed.  The optional
     argument '"symmetric"' is equivalent to not specifying a second
     argument.

     See also: rectwin, bartlett.


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Return the filter coefficients of a Blackman-Harris window of length M.



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blackmannuttall


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 -- Function File: blackmannuttall (M)
 -- Function File: blackmannuttall (M, "periodic")
 -- Function File: blackmannuttall (M, "symmetric")
     Return the filter coefficients of a Blackman-Nuttall window of
     length M.

     If the optional argument '"periodic"' is given, the periodic form
     of the window is returned.  This is equivalent to the window of
     length M+1 with the last coefficient removed.  The optional
     argument '"symmetric"' is equivalent to not specifying a second
     argument.

     See also: nuttallwin, kaiser.


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Return the filter coefficients of a Blackman-Nuttall window of length M.



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bohmanwin


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 -- Function File: bohmanwin (M)
     Return the filter coefficients of a Bohman window of length M.

     See also: rectwin, bartlett.


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Return the filter coefficients of a Bohman window of length M.



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boxcar


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 -- Function File: boxcar (M)
     Return the filter coefficients of a rectangular window of length M.


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Return the filter coefficients of a rectangular window of length M.



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buffer


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 -- Function File: Y = buffer (X, N, P, OPT)
 -- Function File: [Y, Z, OPT] = buffer (...)
     Buffer a signal into a data frame.  The arguments to 'buffer' are

     X
          The data to be buffered.

     N
          The number of rows in the produced data buffer.  This is an
          positive integer value and must be supplied.

     P
          An integer less than N that specifies the under- or overlap
          between column in the data frame.  The default value of P is
          0.

     OPT
          In the case of an overlap, OPT can be either a vector of
          length P or the string 'nodelay'.  If OPT is a vector, then
          the first P entries in Y will be filled with these values.  If
          OPT is the string 'nodelay', then the first value of Y
          corresponds to the first value of X.

          In the can of an underlap, OPT must be an integer between 0
          and '-P'.  The represents the initial underlap of the first
          column of Y.

          The default value for OPT the vector 'zeros (1, P)' in the
          case of an overlap, or 0 otherwise.

     In the case of a single output argument, Y will be padded with
     zeros to fill the missing values in the data frame.  With two
     output arguments Z is the remaining data that has not been used in
     the current data frame.

     Likewise, the output OPT is the overlap, or underlap that might be
     used for a future call to 'code' to allow continuous buffering.


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Buffer a signal into a data frame.



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buttap


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 -- Function File: [Z, P, G] = buttap (N)
     Design lowpass analog Butterworth filter.

     This function exists for MATLAB compatibility only, and is
     equivalent to 'butter (N, 1, "s")'.

     See also: butter.


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Design lowpass analog Butterworth filter.



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butter


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 -- Function File: [B, A] = butter (N, W)
 -- Function File: [B, A] = butter (N, W, "high")
 -- Function File: [B, A] = butter (N, [WL, WH])
 -- Function File: [B, A] = butter (N, [WL, WH], "stop")
 -- Function File: [Z, P, G] = butter (...)
 -- Function File: [A, B, C, D] = butter (...)
 -- Function File: [...] = butter (..., "s")
     Generate a Butterworth filter.  Default is a discrete space (Z)
     filter.

     [b,a] = butter(n, Wc) low pass filter with cutoff pi*Wc radians

     [b,a] = butter(n, Wc, 'high') high pass filter with cutoff pi*Wc
     radians

     [b,a] = butter(n, [Wl, Wh]) band pass filter with edges pi*Wl and
     pi*Wh radians

     [b,a] = butter(n, [Wl, Wh], 'stop') band reject filter with edges
     pi*Wl and pi*Wh radians

     [z,p,g] = butter(...)  return filter as zero-pole-gain rather than
     coefficients of the numerator and denominator polynomials.

     [...]  = butter(...,'s') return a Laplace space filter, W can be
     larger than 1.

     [a,b,c,d] = butter(...)  return state-space matrices

     References:

     Proakis & Manolakis (1992).  Digital Signal Processing.  New York:
     Macmillan Publishing Company.


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Generate a Butterworth filter.



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buttord


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 -- Function File: N = buttord (WP, WS, RP, RS)
 -- Function File: N = buttord ([WP1, WP2], [WS1, WS2], RP, RS)
 -- Function File: [N, WC] = buttord (...)
     Compute the minimum filter order of a Butterworth filter with the
     desired response characteristics.  The filter frequency band edges
     are specified by the passband frequency WP and stopband frequency
     WS.  Frequencies are normalized to the Nyquist frequency in the
     range [0,1].  RP is the allowable passband ripple measured in
     decibels, and RS is the minimum attenuation in the stop band, also
     in decibels.  The output arguments N and WC can be given as inputs
     to 'butter'.

     If WP and WS are scalars, then WP is the passband cutoff frequency
     and WS is the stopband edge frequency.  If WS is greater than WP,
     the filter is a low-pass filter.  If WP is greater than WS, the
     filter is a high-pass filter.

     If WP and WS are vectors of length 2, then WP defines the passband
     interval and WS defines the stopband interval.  If WP is contained
     within WS (WS1 < WP1 < WP2 < WS2), the filter is a band-pass
     filter.  If WS is contained within WP (WP1 < WS1 < WS2 < WP2), the
     filter is a band-stop or band-reject filter.

     Theory: |H(W)|^2 = 1/[1+(W/Wc)^(2N)] = 10^(-R/10) With some
     algebra, you can solve simultaneously for Wc and N given Ws,Rs and
     Wp,Rp.  For high pass filters, subtracting the band edges from
     Fs/2, performing the test, and swapping the resulting Wc back works
     beautifully.  For bandpass and bandstop filters this process
     significantly overdesigns.  Artificially dividing N by 2 in this
     case helps a lot, but it still overdesigns.

     See also: butter, cheb1ord, cheb2ord, ellipord.


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Compute the minimum filter order of a Butterworth filter with the
desired respon



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cceps


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 -- Function File: cceps (X)
 -- Function File: cceps (X, CORRECT)
     Return the complex cepstrum of the vector X.  If the optional
     argument CORRECT has the value 1, a correction method is applied.
     The default is not to do this.


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Return the complex cepstrum of the vector X.



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cheb


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 -- Function File: cheb (N, X)
     Returns the value of the nth-order Chebyshev polynomial calculated
     at the point x.  The Chebyshev polynomials are defined by the
     equations:

                    / cos(n acos(x),    |x| <= 1
            Tn(x) = |
                    \ cosh(n acosh(x),  |x| > 1

     If x is a vector, the output is a vector of the same size, where
     each element is calculated as y(i) = Tn(x(i)).


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Returns the value of the nth-order Chebyshev polynomial calculated at
the point 



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cheb1ap


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 -- Function File: [Z, P, G] = cheb1ap (N, RP)
     Design lowpass analog Chebyshev type I filter.

     This function exists for MATLAB compatibility only, and is
     equivalent to 'cheby1 (N, RP, 1, "s")'.

     See also: cheby1.


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Design lowpass analog Chebyshev type I filter.



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cheb1ord


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 -- Function File: N = cheb1ord (WP, WS, RP, RS)
 -- Function File: N = cheb1ord ([WP1, WP2], [WS1, WS2], RP, RS)
 -- Function File: [N, WC] = cheb1ord (...)
     Compute the minimum filter order of a Chebyshev type I filter with
     the desired response characteristics.  The filter frequency band
     edges are specified by the passband frequency WP and stopband
     frequency WS.  Frequencies are normalized to the Nyquist frequency
     in the range [0,1].  RP is the allowable passband ripple measured
     in decibels, and RS is the minimum attenuation in the stop band,
     also in decibels.  The output arguments N and WC can be given as
     inputs to 'cheby1'.

     If WP and WS are scalars, then WP is the passband cutoff frequency
     and WS is the stopband edge frequency.  If WS is greater than WP,
     the filter is a low-pass filter.  If WP is greater than WS, the
     filter is a high-pass filter.

     If WP and WS are vectors of length 2, then WP defines the passband
     interval and WS defines the stopband interval.  If WP is contained
     within WS (WS1 < WP1 < WP2 < WS2), the filter is a band-pass
     filter.  If WS is contained within WP (WP1 < WS1 < WS2 < WP2), the
     filter is a band-stop or band-reject filter.

     See also: buttord, cheby1, cheb2ord, ellipord.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Compute the minimum filter order of a Chebyshev type I filter with the
desired r



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
cheb2ap


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 233
 -- Function File: [Z, P, G] = cheb2ap (N, RS)
     Design lowpass analog Chebyshev type II filter.

     This function exists for MATLAB compatibility only, and is
     equivalent to 'cheby2 (N, RS, 1, "s")'.

     See also: cheby2.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 47
Design lowpass analog Chebyshev type II filter.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
cheb2ord


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1309
 -- Function File: N = cheb2ord (WP, WS, RP, RS)
 -- Function File: N = cheb2ord ([WP1, WP2], [WS1, WS2], RP, RS)
 -- Function File: [N, WC] = cheb2ord (...)
     Compute the minimum filter order of a Chebyshev type II filter with
     the desired response characteristics.  The filter frequency band
     edges are specified by the passband frequency WP and stopband
     frequency WS.  Frequencies are normalized to the Nyquist frequency
     in the range [0,1].  RP is the allowable passband ripple measured
     in decibels, and RS is the minimum attenuation in the stop band,
     also in decibels.  The output arguments N and WC can be given as
     inputs to 'cheby2'.

     If WP and WS are scalars, then WP is the passband cutoff frequency
     and WS is the stopband edge frequency.  If WS is greater than WP,
     the filter is a low-pass filter.  If WP is greater than WS, the
     filter is a high-pass filter.

     If WP and WS are vectors of length 2, then WP defines the passband
     interval and WS defines the stopband interval.  If WP is contained
     within WS (WS1 < WP1 < WP2 < WS2), the filter is a band-pass
     filter.  If WS is contained within WP (WP1 < WS1 < WS2 < WP2), the
     filter is a band-stop or band-reject filter.

     See also: buttord, cheb1ord, cheby2, ellipord.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Compute the minimum filter order of a Chebyshev type II filter with the
desired 



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
chebwin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1276
 -- Function File: chebwin (M)
 -- Function File: chebwin (M, AT)

     Return the filter coefficients of a Dolph-Chebyshev window of
     length M.  The Fourier transform of the window has a stop-band
     attenuation of AT dB. The default attenuation value is 100 dB.

     For the definition of the Chebyshev window, see

     * Peter Lynch, "The Dolph-Chebyshev Window: A Simple Optimal
     Filter", Monthly Weather Review, Vol.  125, pp.  655-660, April
     1997.  (http://www.maths.tcd.ie/~plynch/Publications/Dolph.pdf)

     * C. Dolph, "A current distribution for broadside arrays which
     optimizes the relationship between beam width and side-lobe level",
     Proc.  IEEE, 34, pp.  335-348.

     The window is described in frequency domain by the expression:

                   Cheb(m-1, beta * cos(pi * k/m))
            W(k) = -------------------------------
                          Cheb(m-1, beta)

     with

            beta = cosh(1/(m-1) * acosh(10^(at/20))

     and Cheb(m,x) denoting the m-th order Chebyshev polynomial
     calculated at the point x.

     Note that the denominator in W(k) above is not computed, and after
     the inverse Fourier transform the window is scaled by making its
     maximum value unitary.

     See also: kaiser.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 71
Return the filter coefficients of a Dolph-Chebyshev window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
cheby1


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1191
 -- Function File: [B, A] = cheby1 (N, RP, W)
 -- Function File: [B, A] = cheby1 (N, RP, W, "high")
 -- Function File: [B, A] = cheby1 (N, RP, [WL, WH])
 -- Function File: [B, A] = cheby1 (N, RP, [WL, WH], "stop")
 -- Function File: [Z, P, G] = cheby1 (...)
 -- Function File: [A, B, C, D] = cheby1 (...)
 -- Function File: [...] = cheby1 (..., "s")
     Generate a Chebyshev type I filter with RP dB of passband ripple.

     [b, a] = cheby1(n, Rp, Wc) low pass filter with cutoff pi*Wc
     radians

     [b, a] = cheby1(n, Rp, Wc, 'high') high pass filter with cutoff
     pi*Wc radians

     [b, a] = cheby1(n, Rp, [Wl, Wh]) band pass filter with edges pi*Wl
     and pi*Wh radians

     [b, a] = cheby1(n, Rp, [Wl, Wh], 'stop') band reject filter with
     edges pi*Wl and pi*Wh radians

     [z, p, g] = cheby1(...)  return filter as zero-pole-gain rather
     than coefficients of the numerator and denominator polynomials.

     [...]  = cheby1(...,'s') return a Laplace space filter, W can be
     larger than 1.

     [a,b,c,d] = cheby1(...)  return state-space matrices

     References:

     Parks & Burrus (1987).  Digital Filter Design.  New York: John
     Wiley & Sons, Inc.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 65
Generate a Chebyshev type I filter with RP dB of passband ripple.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
cheby2


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1204
 -- Function File: [B, A] = cheby2 (N, RS, WC)
 -- Function File: [B, A] = cheby2 (N, RS, WC, "high")
 -- Function File: [B, A] = cheby2 (N, RS, [WL, WH])
 -- Function File: [B, A] = cheby2 (N, RS, [WL, WH], "stop")
 -- Function File: [Z, P, G] = cheby2 (...)
 -- Function File: [A, B, C, D] = cheby2 (...)
 -- Function File: [...] = cheby2 (..., "s")
     Generate a Chebyshev type II filter with RS dB of stopband
     attenuation.

     [b, a] = cheby2(n, Rs, Wc) low pass filter with cutoff pi*Wc
     radians

     [b, a] = cheby2(n, Rs, Wc, 'high') high pass filter with cutoff
     pi*Wc radians

     [b, a] = cheby2(n, Rs, [Wl, Wh]) band pass filter with edges pi*Wl
     and pi*Wh radians

     [b, a] = cheby2(n, Rs, [Wl, Wh], 'stop') band reject filter with
     edges pi*Wl and pi*Wh radians

     [z, p, g] = cheby2(...)  return filter as zero-pole-gain rather
     than coefficients of the numerator and denominator polynomials.

     [...]  = cheby2(...,'s') return a Laplace space filter, W can be
     larger than 1.

     [a,b,c,d] = cheby2(...)  return state-space matrices

     References:

     Parks & Burrus (1987).  Digital Filter Design.  New York: John
     Wiley & Sons, Inc.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 71
Generate a Chebyshev type II filter with RS dB of stopband attenuation.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
chirp


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1084
 -- Function File: chirp (T)
 -- Function File: chirp (T, F0)
 -- Function File: chirp (T, F0, T1)
 -- Function File: chirp (T, F0, T1, F1)
 -- Function File: chirp (T, F0, T1, F1, FORM)
 -- Function File: chirp (T, F0, T1, F1, FORM, PHASE)

     Evaluate a chirp signal at time T.  A chirp signal is a frequency
     swept cosine wave.

     T
          vector of times to evaluate the chirp signal
     F0
          frequency at time t=0 [ 0 Hz ]
     T1
          time t1 [ 1 sec ]
     F1
          frequency at time t=t1 [ 100 Hz ]
     FORM
          shape of frequency sweep 'linear' f(t) = (f1-f0)*(t/t1) + f0
          'quadratic' f(t) = (f1-f0)*(t/t1)^2 + f0 'logarithmic' f(t) =
          (f1-f0)^(t/t1) + f0
     PHASE
          phase shift at t=0

     Example specgram(chirp([0:0.001:5])); # linear, 0-100Hz in 1 sec
     specgram(chirp([-2:0.001:15], 400, 10, 100, 'quadratic'));
     soundsc(chirp([0:1/8000:5], 200, 2, 500, "logarithmic"),8000);

     If you want a different sweep shape f(t), use the following: y =
     cos(2*pi*integral(f(t)) + 2*pi*f0*t + phase);


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 34
Evaluate a chirp signal at time T.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 14
clustersegment


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 710
 -- Function File: CLUSTERIDX = clustersegment (UNOS)
     Calculate boundary indexes of clusters of 1's.

     The function calculates the initial index and end index of the
     sequences of 1's in the rows of UNOS.  The clusters are sought in
     the rows of the array UNOS.

     The result is returned in a cell array of size 1-by-NP, where NP is
     the number of rows in UNOS.  Each element of the cell has two rows.
     The first row is the initial index of a sequence of 1's and the
     second row is the end index of that sequence.

     If NP == 1 the output is a matrix with two rows.

     The function works by finding the indexes of jumps between
     consecutive values in the rows of UNOS.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 46
Calculate boundary indexes of clusters of 1's.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
cmorwavf


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 99
 -- Function File: [PSI, X] = cmorwavf (LB, UB, N, FB, FC)
     Compute the Complex Morlet wavelet.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 35
Compute the Complex Morlet wavelet.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
cohere


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 377
 Usage:
   [Pxx,freq] = cohere(x,y,Nfft,Fs,window,overlap,range,plot_type,detrend)

     Estimate (mean square) coherence of signals "x" and "y".
     Use the Welch (1967) periodogram/FFT method.
     Compatible with Matlab R11 cohere and earlier.
     See "help pwelch" for description of arguments, hints and references
     --- especially hint (7) for Matlab R11 defaults.




# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
 Usage:
   [Pxx,freq] = cohere(x,y,Nfft,Fs,window,overlap,range,plot_type,detren



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
convmtx


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 494
 -- Function File: convmtx (A, N)
     If A is a column vector and X is a column vector of length N, then

     'convmtx(A, N) * X'

     gives the convolution of of A and X and is the same as 'conv(A,
     X)'.  The difference is if many vectors are to be convolved with
     the same vector, then this technique is possibly faster.

     Similarly, if A is a row vector and X is a row vector of length N,
     then

     'X * convmtx(A, N)'

     is the same as 'conv(X, A)'.

See also: conv.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 66
If A is a column vector and X is a column vector of length N, then



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
cplxreal


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 699
 -- Function File: [ZC, ZR] = cplxreal (Z, THRESH)
     Split the vector z into its complex (ZC) and real (ZR) elements,
     eliminating one of each complex-conjugate pair.

     INPUTS:
        * Z = row- or column-vector of complex numbers
        * THRESH = tolerance threshold for numerical comparisons
          (default = 100*eps)

     RETURNED:
        * ZC = elements of Z having positive imaginary parts
        * ZR = elements of Z having zero imaginary part

     Each complex element of Z is assumed to have a complex-conjugate
     counterpart elsewhere in Z as well.  Elements are declared real if
     their imaginary parts have magnitude less than THRESH.

     See also: cplxpair.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Split the vector z into its complex (ZC) and real (ZR) elements,
eliminating one



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
cpsd


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 503
 -- Function File: [PXX, FREQ] = cpsd (X, Y)
 -- Function File: [...] = cpsd (X, Y, WINDOW)
 -- Function File: [...] = cpsd (X, Y, WINDOW, OVERLAP)
 -- Function File: [...] = cpsd (X, Y, WINDOW, OVERLAP, NFFT)
 -- Function File: [...] = cpsd (X, Y, WINDOW, OVERLAP, NFFT, FS)
 -- Function File: [...] = cpsd (X, Y, WINDOW, OVERLAP, NFFT, FS, RANGE)
 -- Function File: cpsd (...)

     Estimate cross power spectrum of data X and Y by the Welch (1967)
     periodogram/FFT method.

     See also: pwelch.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Estimate cross power spectrum of data X and Y by the Welch (1967)
periodogram/FF



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3
csd


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 358
 Usage:
   [Pxx,freq] = csd(x,y,Nfft,Fs,window,overlap,range,plot_type,detrend)

     Estimate cross power spectrum of data "x" and "y" by the Welch (1967)
     periodogram/FFT method.  Compatible with Matlab R11 csd and earlier.
     See "help pwelch" for description of arguments, hints and references
     --- especially hint (7) for Matlab R11 defaults.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 79
 Usage:
   [Pxx,freq] = csd(x,y,Nfft,Fs,window,overlap,range,plot_type,detrend)



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3
czt


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 978
 -- Function File: czt (X)
 -- Function File: czt (X, M)
 -- Function File: czt (X, M, W)
 -- Function File: czt (X, M, W, A)
     Chirp z-transform.  Compute the frequency response starting at a
     and stepping by w for m steps.  a is a point in the complex plane,
     and w is the ratio between points in each step (i.e., radius
     increases exponentially, and angle increases linearly).

     To evaluate the frequency response for the range f1 to f2 in a
     signal with sampling frequency Fs, use the following:

          m = 32;                               ## number of points desired
          w = exp(-j*2*pi*(f2-f1)/((m-1)*Fs));  ## freq. step of f2-f1/m
          a = exp(j*2*pi*f1/Fs);                ## starting at frequency f1
          y = czt(x, m, w, a);

     If you don't specify them, then the parameters default to a Fourier
     transform: m=length(x), w=exp(-j*2*pi/m), a=1

     If x is a matrix, the transform will be performed column-by-column.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 18
Chirp z-transform.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
data2fun


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1111
 -- Function File: [FHANDLE, FULLNAME] = data2fun (TI, YI)
 -- Function File: [...] = data2fun (TI, YI,PROPERTY,VALUE)
     Creates a vectorized function based on data samples using
     interpolation.

     The values given in YI (N-by-k matrix) correspond to evaluations of
     the function y(t) at the points TI (N-by-1 matrix).  The data is
     interpolated and the function handle to the generated interpolant
     is returned.

     The function accepts property-value pairs described below.

     'file'
          Code is generated and .m file is created.  The VALUE contains
          the name of the function.  The returned function handle is a
          handle to that file.  If VALUE is empty, then a name is
          automatically generated using 'tmpnam' and the file is created
          in the current directory.  VALUE must not have an extension,
          since .m will be appended.  Numerical value used in the
          function are stored in a .mat file with the same name as the
          function.

     'interp'
          Type of interpolation.  See 'interp1'.

     See also: interp1.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 72
Creates a vectorized function based on data samples using interpolation.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3
dct


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 779
 -- Function File: dct (X)
 -- Function File: dct (X, N)
     Compute the discrete cosine transform of X.  If N is given, then X
     is padded or trimmed to length N before computing the transform.
     If X is a matrix, compute the transform along the columns of the
     the matrix.  The transform is faster if X is real-valued and has
     even length.

     The discrete cosine transform X can be defined as follows:

                        N-1
            X[k] = w(k) sum x[n] cos (pi (2n+1) k / 2N ),  k = 0, ..., N-1
                        n=0

     with w(0) = sqrt(1/N) and w(k) = sqrt(2/N), k = 1, ..., N-1.  There
     are other definitions with different scaling of X[k], but this form
     is common in image processing.

     See also: idct, dct2, idct2, dctmtx.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 43
Compute the discrete cosine transform of X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
dct2


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 275
 -- Function File: dct2 (X)
 -- Function File: dct2 (X, M, N)
 -- Function File: dct2 (X, [M, N])
     Compute the 2-D discrete cosine transform of matrix X.  If M and N
     are specified, the input is padded or trimmed to the desired size.

     See also: dct, idct, idct2.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 54
Compute the 2-D discrete cosine transform of matrix X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
dctmtx


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 670
 -- Function File: dctmtx (N)
     Return the DCT transformation matrix of size N-by-N.

     If A is an N-by-N matrix, then the following are true:

              T*A    == dct(A),  T'*A   == idct(A)
              T*A*T' == dct2(A), T'*A*T == idct2(A)

     A DCT transformation matrix is useful for doing things like jpeg
     image compression, in which an 8x8 DCT matrix is applied to
     non-overlapping blocks throughout an image and only a subblock on
     the top left of each block is kept.  During restoration, the
     remainder of the block is filled with zeros and the inverse
     transform is applied to the block.

     See also: dct, idct, dct2, idct2.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 52
Return the DCT transformation matrix of size N-by-N.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
decimate


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1381
 -- Function File: Y = decimate (X, Q)
 -- Function File: Y = decimate (X, Q, N)
 -- Function File: Y = decimate (..., "fir")

     Downsample the signal X by a reduction factor of Q.  A lowpass
     antialiasing filter is applied to the signal prior to reducing the
     input sequence.  By default, an order N Chebyshev type I filter is
     used.  If N is not specified, the default is 8.

     If the optional argument '"fir"' is given, an order N FIR filter is
     used, with a default order of 30 if N is not given.

     Note that Q must be an integer for this rate change method.

     Example:
          ## Generate a signal that starts away from zero, is slowly varying
          ## at the start and quickly varying at the end, decimate and plot.
          ## Since it starts away from zero, you will see the boundary
          ## effects of the antialiasing filter clearly.  Next you will see
          ## how it follows the curve nicely in the slowly varying early
          ## part of the signal, but averages the curve in the quickly
          ## varying late part of the signal.
          t = 0:0.01:2;
          x = chirp (t, 2, .5, 10, "quadratic") + sin (2*pi*t*0.4);
          y = decimate (x, 4);
          stem (t(1:121) * 1000, x(1:121), "-g;Original;"); hold on; # original
          stem (t(1:4:121) * 1000, y(1:31), "-r;Decimated;"); hold off; # decimated


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 51
Downsample the signal X by a reduction factor of Q.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
dftmtx


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 410
 -- Function File: D = dftmtx (N)
     Compute the N-by-N Fourier transformation matrix.  This is the
     matrix D such that the Fourier transform of a column vector of
     length N is given by 'dftmtx(N) * X' and the inverse Fourier
     transform is given by 'inv(dftmtx(N)) * X'.

     In general this is less efficient than calling the 'fft' and 'ifft'
     functions directly.

     See also: fft, ifft.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 49
Compute the N-by-N Fourier transformation matrix.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 13
digitrevorder


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 330
 -- Function File: Y = digitrevorder (X, R)
 -- Function File: [Y, I] = digitrevorder (X, R)
     Reorder the elements of the vector X in digit-reversed order.  The
     elements of X are converted to radix R and reversed.  The reordered
     indices of the elements of X are returned in I.

     See also: bitrevorder, fft, ifft.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 61
Reorder the elements of the vector X in digit-reversed order.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
diric


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 113
 -- Function File: Y = diric (X,N)
     Compute the dirichlet function.

     See also: sinc, gauspuls, sawtooth.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 31
Compute the dirichlet function.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 10
downsample


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 509
 -- Function File: Y = downsample (X, N)
 -- Function File: Y = downsample (X, N, OFFSET)
     Downsample the signal, selecting every Nth element.  If X is a
     matrix, downsample every column.

     For most signals you will want to use 'decimate' instead since it
     prefilters the high frequency components of the signal and avoids
     aliasing effects.

     If OFFSET is defined, select every Nth element starting at sample
     OFFSET.

     See also: decimate, interp, resample, upfirdn, upsample.


# name: <cell-element>
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# elements: 1
# length: 51
Downsample the signal, selecting every Nth element.



# name: <cell-element>
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# elements: 1
# length: 3
dst


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 480
 -- Function File: Y = dst (X)
 -- Function File: Y = dst (X, N)
     Computes the type I discrete sine transform of X.  If N is given,
     then X is padded or trimmed to length N before computing the
     transform.  If X is a matrix, compute the transform along the
     columns of the the matrix.

     The discrete sine transform X of x can be defined as follows:

            N
     X[k] = sum x[n] sin (pi n k / (N+1) ),  k = 1, ..., N
            n=1

     See also: idst.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 49
Computes the type I discrete sine transform of X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3
dwt


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 522
 -- Function File: [U, V] = dwt (X, WNAME)
 -- Function File: [U, V] = dwt (X, HP, GP)
 -- Function File: [U, V] = dwt (X, HP, GP, ...)
     Discrete wavelet transform (1D).

     *Inputs*
     X
          Signal vector.
     WNAME
          Wavelet name.
     HP
          Coefficients of low-pass decomposition FIR filter.
     GP
          Coefficients of high-pass decomposition FIR filter.

     *Outputs*
     U
          Signal vector of average, approximation.
     V
          Signal vector of difference, detail.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 32
Discrete wavelet transform (1D).



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
ellip


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1442
 -- Function File: [B, A] = ellip (N, RP, RS, WP)
 -- Function File: [B, A] = ellip (N, RP, RS, WP, "high")
 -- Function File: [B, A] = ellip (N, RP, RS, [WL, WH])
 -- Function File: [B, A] = ellip (N, RP, RS, [WL, WH], "stop")
 -- Function File: [Z, P, G] = ellip (...)
 -- Function File: [A, B, C, D] = ellip (...)
 -- Function File: [...] = ellip (..., "s")

     Generate an elliptic or Cauer filter with RP dB of passband ripple
     and RS dB of stopband attenuation.

     [b,a] = ellip(n, Rp, Rs, Wp) low pass filter with order n, cutoff
     pi*Wp radians, Rp decibels of ripple in the passband and a stopband
     Rs decibels down.

     [b,a] = ellip(n, Rp, Rs, Wp, 'high') high pass filter with cutoff
     pi*Wp...

     [b,a] = ellip(n, Rp, Rs, [Wl, Wh]) band pass filter with band pass
     edges pi*Wl and pi*Wh ...

     [b,a] = ellip(n, Rp, Rs, [Wl, Wh], 'stop') band reject filter with
     edges pi*Wl and pi*Wh, ...

     [z,p,g] = ellip(...)  return filter as zero-pole-gain.

     [...]  = ellip(...,'s') return a Laplace space filter, W can be
     larger than 1.

     [a,b,c,d] = ellip(...)  return state-space matrices

     References:

     - Oppenheim, Alan V., Discrete Time Signal Processing, Hardcover,
     1999.  - Parente Ribeiro, E., Notas de aula da disciplina TE498 -
     Processamento Digital de Sinais, UFPR, 2001/2002.  - Kienzle, Paul,
     functions from Octave-Forge, 1999 (http://octave.sf.net).


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Generate an elliptic or Cauer filter with RP dB of passband ripple and
RS dB of 



# name: <cell-element>
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# elements: 1
# length: 7
ellipap


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 230
 -- Function File: [Z, P, G] = ellipap (N, RP, RS)
     Design lowpass analog elliptic filter.

     This function exists for MATLAB compatibility only, and is
     equivalent to 'ellip (N, RP, RS, 1, "s")'.

     See also: ellip.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 38
Design lowpass analog elliptic filter.



# name: <cell-element>
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# elements: 1
# length: 8
ellipord


# name: <cell-element>
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# elements: 1
# length: 1426
 -- Function File: N = ellipord (WP, WS, RP, RS)
 -- Function File: N = ellipord ([WP1, WP2], [WS1, WS2], RP, RS)
 -- Function File: [N, WC] = ellipord (...)
     Compute the minimum filter order of an elliptic filter with the
     desired response characteristics.  The filter frequency band edges
     are specified by the passband frequency WP and stopband frequency
     WS.  Frequencies are normalized to the Nyquist frequency in the
     range [0,1].  RP is the allowable passband ripple measured in
     decibels, and RS is the minimum attenuation in the stop band, also
     in decibels.  The output arguments N and WC can be given as inputs
     to 'ellip'.

     If WP and WS are scalars, then WP is the passband cutoff frequency
     and WS is the stopband edge frequency.  If WS is greater than WP,
     the filter is a low-pass filter.  If WP is greater than WS, the
     filter is a high-pass filter.

     If WP and WS are vectors of length 2, then WP defines the passband
     interval and WS defines the stopband interval.  If WP is contained
     within WS (WS1 < WP1 < WP2 < WS2), the filter is a band-pass
     filter.  If WS is contained within WP (WP1 < WS1 < WS2 < WP2), the
     filter is a band-stop or band-reject filter.

     Reference: Lamar, Marcus Vinicius, 'Notas de aula da disciplina TE
     456 - Circuitos Analogicos II', UFPR, 2001/2002.

     See also: buttord, cheb1ord, cheb2ord, ellip.


# name: <cell-element>
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# elements: 1
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Compute the minimum filter order of an elliptic filter with the desired
response



# name: <cell-element>
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# elements: 1
# length: 3
fht


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 776
 -- Function File: M = fht (D)
 -- Function File: M = fht (D, N)
 -- Function File: M = fht (D, N, DIM)
     Calculate the Fast Hartley Transform of real input D.  If D is a
     matrix, the Hartley transform is calculated along the columns by
     default.  The options N and DIM are similar to the options of FFT
     function.

     The forward and inverse Hartley transforms are the same (except for
     a scale factor of 1/N for the inverse Hartley transform), but
     implemented using different functions.

     The definition of the forward hartley transform for vector d, m[K]
     = \sum_{i=0}^{N-1} d[i]*(cos[K*2*pi*i/N] + sin[K*2*pi*i/N]), for 0
     <= K < N. m[K] = \sum_{i=0}^{N-1} d[i]*CAS[K*i], for 0 <= K < N.

          fht(1:4)

     See also: ifht, fft.


# name: <cell-element>
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# length: 53
Calculate the Fast Hartley Transform of real input D.



# name: <cell-element>
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# elements: 1
# length: 8
filtfilt


# name: <cell-element>
# type: sq_string
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 -- Function File: Y = filtfilt (B, A, X)

     Forward and reverse filter the signal.  This corrects for phase
     distortion introduced by a one-pass filter, though it does square
     the magnitude response in the process.  That's the theory at least.
     In practice the phase correction is not perfect, and magnitude
     response is distorted, particularly in the stop band.

     Example
          [b, a]=butter(3, 0.1);                  # 5 Hz low-pass filter
          t = 0:0.01:1.0;                         # 1 second sample
          x=sin(2*pi*t*2.3)+0.25*randn(size(t));  # 2.3 Hz sinusoid+noise
          y = filtfilt(b,a,x); z = filter(b,a,x); # apply filter
          plot(t,x,';data;',t,y,';filtfilt;',t,z,';filter;')


# name: <cell-element>
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# length: 38
Forward and reverse filter the signal.



# name: <cell-element>
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# elements: 1
# length: 6
filtic


# name: <cell-element>
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# elements: 1
# length: 847
 -- Function File: ZF = filtic (B, A, Y)
 -- Function File: ZF = filtic (B, A, Y, X)

     Set initial condition vector for filter function The vector zf has
     the same values that would be obtained from function filter given
     past inputs x and outputs y

     The vectors x and y contain the most recent inputs and outputs
     respectively, with the newest values first:

     x = [x(-1) x(-2) ...  x(-nb)], nb = length(b)-1 y = [y(-1) y(-2)
     ...  y(-na)], na = length(a)-a

     If length(x)<nb then it is zero padded If length(y)<na then it is
     zero padded

     zf = filtic(b, a, y) Initial conditions for filter with
     coefficients a and b and output vector y, assuming input vector x
     is zero

     zf = filtic(b, a, y, x) Initial conditions for filter with
     coefficients a and b input vector x and output vector y


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Set initial condition vector for filter function The vector zf has the
same valu



# name: <cell-element>
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# elements: 1
# length: 9
findpeaks


# name: <cell-element>
# type: sq_string
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# length: 2797
 -- Function File: [PKS, LOC, EXTRA] = findpeaks (DATA)
 -- Function File: ... = findpeaks (..., PROPERTY, VALUE)
 -- Function File: ... = findpeaks (..., "DoubleSided")
     Finds peaks on DATA.

     Peaks of a positive array of data are defined as local maxima.  For
     double-sided data, they are maxima of the positive part and minima
     of the negative part.  DATA is expected to be a single column
     vector.

     The function returns the value of DATA at the peaks in PKS.  The
     index indicating their position is returned in LOC.

     The third output argument is a structure with additional
     information:

     "parabol"
          A structure containing the parabola fitted to each returned
          peak.  The structure has two fields, "x" and "pp".  The field
          "pp" contains the coefficients of the 2nd degree polynomial
          and "x" the extrema of the intercal here it was fitted.

     "height"
          The estimated height of the returned peaks (in units of DATA).

     "baseline"
          The height at which the roots of the returned peaks were
          calculated (in units of DATA).

     "roots"
          The abscissa values (in index units) at which the parabola
          fitted to each of the returned peaks crosses the "baseline"
          value.  The width of the peak is calculated by 'diff(roots)'.

     This function accepts property-value pair given in the list below:

     "MinPeakHeight"
          Minimum peak height (positive scalar).  Only peaks that exceed
          this value will be returned.  For data taking positive and
          negative values use the option "DoubleSided".  Default value
          '2*std (abs (detrend (data,0)))'.

     "MinPeakDistance"
          Minimum separation between (positive integer).  Peaks
          separated by less than this distance are considered a single
          peak.  This distance is also used to fit a second order
          polynomial to the peaks to estimate their width, therefore it
          acts as a smoothing parameter.  Default value 4.

     "MinPeakWidth"
          Minimum width of peaks (positive integer).  The width of the
          peaks is estimated using a parabola fitted to the neighborhood
          of each peak.  The neighborhood size is equal to the value of
          "MinPeakDistance".  The width is evaluated at the half height
          of the peak with baseline at "MinPeakHeight".  Default value
          2.

     "DoubleSided"
          Tells the function that data takes positive and negative
          values.  The base-line for the peaks is taken as the mean
          value of the function.  This is equivalent as passing the
          absolute value of the data after removing the mean.

     Run 'demo findpeaks' to see some examples.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 20
Finds peaks on DATA.



# name: <cell-element>
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# elements: 1
# length: 4
fir1


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1553
 -- Function File: B = fir1 (N, W)
 -- Function File: B = fir1 (N, W, TYPE)
 -- Function File: B = fir1 (N, W, TYPE, WINDOW)
 -- Function File: B = fir1 (N, W, TYPE, WINDOW, NOSCALE)

     Produce an order N FIR filter with the given frequency cutoff W,
     returning the N+1 filter coefficients in B.  If W is a scalar, it
     specifies the frequency cutoff for a lowpass or highpass filter.
     If W is a two-element vector, the two values specify the edges of a
     bandpass or bandstop filter.  If W is an N-element vector, each
     value specifies a band edge of a multiband pass/stop filter.

     The filter TYPE can be specified with one of the following strings:
     "low", "high", "stop", "pass", "bandpass", "DC-0", or "DC-1".  The
     default is "low" is W is a scalar, "pass" if W is a pair, or "DC-0"
     if W is a vector with more than 2 elements.

     An optional shaping WINDOW can be given as a vector with length
     N+1.  If not specified, a Hamming window of length N+1 is used.

     With the option "noscale", the filter coefficients are not
     normalized.  The default is to normalize the filter such that the
     magnitude response of the center of the first passband is 1.

     To apply the filter, use the return vector B with the 'filter'
     function, for example 'y = filter (b, 1, x)'.

     Examples:
          freqz (fir1 (40, 0.3));
          freqz (fir1 (15, [0.2, 0.5], "stop"));  # note the zero-crossing at 0.1
          freqz (fir1 (15, [0.2, 0.5], "stop", "noscale"));

     See also: filter, fir2.


# name: <cell-element>
# type: sq_string
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# length: 80
Produce an order N FIR filter with the given frequency cutoff W,
returning the N



# name: <cell-element>
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# elements: 1
# length: 4
fir2


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1672
 -- Function File: B = fir2 (N, F, M)
 -- Function File: B = fir2 (N, F, M, GRID_N)
 -- Function File: B = fir2 (N, F, M, GRID_N, RAMP_N)
 -- Function File: B = fir2 (N, F, M, GRID_N, RAMP_N, WINDOW)

     Produce an order N FIR filter with arbitrary frequency response M
     over frequency bands F, returning the N+1 filter coefficients in B.
     The vector F specifies the frequency band edges of the filter
     response and M specifies the magnitude response at each frequency.

     The vector F must be nondecreasing over the range [0,1], and the
     first and last elements must be 0 and 1, respectively.  A
     discontinuous jump in the frequency response can be specified by
     duplicating a band edge in F with different values in M.

     The resolution over which the frequency response is evaluated can
     be controlled with the GRID_N argument.  The default is 512 or the
     next larger power of 2 greater than the filter length.

     The band transition width for discontinuities can be controlled
     with the RAMP_N argument.  The default is GRID_N/25.  Larger values
     will result in wider band transitions but better stopband
     rejection.

     An optional shaping WINDOW can be given as a vector with length
     N+1.  If not specified, a Hamming window of length N+1 is used.

     To apply the filter, use the return vector B with the 'filter'
     function, for example 'y = filter (b, 1, x)'.

     Example:
          f = [0, 0.3, 0.3, 0.6, 0.6, 1]; m = [0, 0, 1, 1/2, 0, 0];
          [h, w] = freqz (fir2 (100, f, m));
          plot (f, m, ";target response;", w/pi, abs (h), ";filter response;");

     See also: filter, fir1.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Produce an order N FIR filter with arbitrary frequency response M over
frequency



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
firls


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1113
 -- Function File: B = firls (N, F, A)
 -- Function File: B = firls (N, F, A, W)

     FIR filter design using least squares method.  Returns a length N+1
     linear phase filter such that the integral of the weighted mean
     squared error in the specified bands is minimized.

     The vector F specifies the frequencies of the band edges,
     normalized so that half the sample frequency is equal to 1.  Each
     band is specified by two frequencies, to the vector must have an
     even length.

     The vector A specifies the amplitude of the desired response at
     each band edge.

     The optional argument W is a weighting function that contains one
     value for each band that weights the mean squared error in that
     band.

     A must be the same length as F, and W must be half the length of F.
     N must be even.  If given an odd value, 'firls' increments it by 1.

     The least squares optimization algorithm for computing FIR filter
     coefficients is derived in detail in:

     I. Selesnick, "Linear-Phase FIR Filter Design by Least Squares,"
     http://cnx.org/content/m10577


# name: <cell-element>
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# length: 45
FIR filter design using least squares method.



# name: <cell-element>
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# elements: 1
# length: 10
flattopwin


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 -- Function File: flattopwin (M)
 -- Function File: flattopwin (M, "periodic")
 -- Function File: flattopwin (M, "symmetric")

     Return the filter coefficients of a Flat Top window of length M.
     The Flat Top window is defined by the function f(w):

            f(w) = 1 - 1.93 cos(2 pi w) + 1.29 cos(4 pi w)
                     - 0.388 cos(6 pi w) + 0.0322cos(8 pi w)

     where w = i/(m-1) for i=0:m-1 for a symmetric window, or w = i/m
     for i=0:m-1 for a periodic window.  The default is symmetric.  The
     returned window is normalized to a peak of 1 at w = 0.5.

     This window has low pass-band ripple, but high bandwidth.

     According to [1]:

     The main use for the Flat Top window is for calibration, due to its
     negligible amplitude errors.

     [1] Gade, S; Herlufsen, H; (1987) "Use of weighting functions in
     DFT/FFT analysis (Part I)", Bruel & Kjaer Technical Review No.3.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 64
Return the filter coefficients of a Flat Top window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 9
fracshift


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 287
 -- Function File: [Y, H] = fracshift (X, D)
 -- Function File: Y = fracshift (X, D, H)
     Shift the series X by a (possibly fractional) number of samples D.
     The interpolator H is either specified or either designed with a
     Kaiser-windowed sinecard.

     See also: circshift.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 66
Shift the series X by a (possibly fractional) number of samples D.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
freqs


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 385
 -- Function File: H = freqs (B, A, W)
 -- Function File: freqs (B, A, W)

     Compute the s-plane frequency response of the IIR filter B(s)/A(s)
     as H = polyval(B,j*W)./polyval(A,j*W). If called with no output
     argument, a plot of magnitude and phase are displayed.

     Example:
          b = [1 2]; a = [1 1];
          w = linspace (0, 4, 128);
          freqs (b, a, w);


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Compute the s-plane frequency response of the IIR filter B(s)/A(s) as H
= polyva



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 10
freqs_plot


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 87
 -- Function File: freqs_plot (W, H)
     Plot the amplitude and phase of the vector H.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 45
Plot the amplitude and phase of the vector H.



# name: <cell-element>
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# elements: 1
# length: 4
fwhm


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# length: 1264
 -- Function File: F = fwhm (Y)
 -- Function File: F = fwhm (X, Y)
 -- Function File: F = fwhm (..., "zero")
 -- Function File: F = fwhm (..., "min")
 -- Function File: F = fwhm (..., "alevel", LEVEL)
 -- Function File: F = fwhm (..., "rlevel", LEVEL)

     Compute peak full-width at half maximum (FWHM) or at another level
     of peak maximum for vector or matrix data Y, optionally sampled as
     y(x).  If Y is a matrix, return FWHM for each column as a row
     vector.

     The default option "zero" computes fwhm at half maximum, i.e.
     0.5*max(y).  The option "min" computes fwhm at the middle curve,
     i.e.  0.5*(min(y)+max(y)).

     The option "rlevel" computes full-width at the given relative level
     of peak profile, i.e.  at rlevel*max(y) or rlevel*(min(y)+max(y)),
     respectively.  For example, 'fwhm (..., "rlevel", 0.1)' computes
     full width at 10 % of peak maximum with respect to zero or minimum;
     FWHM is equivalent to 'fwhm(..., "rlevel", 0.5)'.

     The option "alevel" computes full-width at the given absolute level
     of Y.

     Return 0 if FWHM does not exist (e.g.  monotonous function or the
     function does not cut horizontal line at rlevel*max(y) or
     rlevel*(max(y)+min(y)) or alevel, respectively).


# name: <cell-element>
# type: sq_string
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# length: 80
Compute peak full-width at half maximum (FWHM) or at another level of
peak maxim



# name: <cell-element>
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# elements: 1
# length: 4
fwht


# name: <cell-element>
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# length: 981
 -- Function File: fwht (X)
 -- Function File: fwht (X, N)
 -- Function File: fwht (X, N, ORDER)
     Compute the Walsh-Hadamard transform of X using the Fast
     Walsh-Hadamard Transform (FWHT) algorithm.  If the input is a
     matrix, the FWHT is calculated along the columns of X.

     The number of elements of X must be a power of 2; if not, the input
     will be extended and filled with zeros.  If a second argument is
     given, the input is truncated or extended to have length N.

     The third argument specifies the ORDER in which the returned
     Walsh-Hadamard transform coefficients should be arranged.  The
     ORDER may be any of the following strings:

     "sequency"
          The coefficients are returned in sequency order.  This is the
          default if ORDER is not given.

     "hadamard"
          The coefficients are returned in Hadamard order.

     "dyadic"
          The coefficients are returned in Gray code order.

     See also: ifwht.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Compute the Walsh-Hadamard transform of X using the Fast Walsh-Hadamard
Transfor



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
gauspuls


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 94
 -- Function File: Y = gauspuls (T,FC,BW)
     Return the Gaussian modulated sinusoidal pulse.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 47
Return the Gaussian modulated sinusoidal pulse.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
gaussian


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 560
 -- Function File: gaussian (M)
 -- Function File: gaussian (M, A)

     Return a Gaussian convolution window of length M.  The width of the
     window is inversely proportional to the parameter A.  Use larger A
     for a narrower window.  Use larger M for longer tails.

     w = exp ( -(a*x)^2/2 )

     for x = linspace ( -(m-1)/2, (m-1)/2, m ).

     Width a is measured in frequency units (sample rate/num samples).
     It should be f when multiplying in the time domain, but 1/f when
     multiplying in the frequency domain (for use in convolutions).


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 49
Return a Gaussian convolution window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
gausswin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 367
 -- Function File: gausswin (M)
 -- Function File: gausswin (M, A)

     Return the filter coefficients of a Gaussian window of length M.
     The width of the window is inversely proportional to the parameter
     A.  Use larger A for a narrow window.  Use larger M for a smoother
     curve.

     w = exp ( -(a*x)^2/2 )

     for x = linspace(-(m-1)/m, (m-1)/m, m)


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 64
Return the filter coefficients of a Gaussian window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 9
gmonopuls


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 75
 -- Function File: Y = gmonopuls (T,FC)
     Return the gaussian monopulse.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 30
Return the gaussian monopulse.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
grpdelay


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3129
 -- Function File: [G, W] = grpdelay (B)
 -- Function File: [G, W] = grpdelay (B, A)
 -- Function File: [G, W] = grpdelay (..., N)
 -- Function File: [G, W] = grpdelay (..., N, "whole")
 -- Function File: [G, F] = grpdelay (..., N, FS)
 -- Function File: [G, F] = grpdelay (..., N, "whole", FS)
 -- Function File: [G, W] = grpdelay (..., W)
 -- Function File: [G, F] = grpdelay (..., F, FS)
 -- Function File: grpdelay (...)
     Compute the group delay of a filter.

     [g, w] = grpdelay(b) returns the group delay g of the FIR filter
     with coefficients b.  The response is evaluated at 512 angular
     frequencies between 0 and pi.  w is a vector containing the 512
     frequencies.  The group delay is in units of samples.  It can be
     converted to seconds by multiplying by the sampling period (or
     dividing by the sampling rate fs).

     [g, w] = grpdelay(b,a) returns the group delay of the rational IIR
     filter whose numerator has coefficients b and denominator
     coefficients a.

     [g, w] = grpdelay(b,a,n) returns the group delay evaluated at n
     angular frequencies.  For fastest computation n should factor into
     a small number of small primes.

     [g, w] = grpdelay(b,a,n,'whole') evaluates the group delay at n
     frequencies between 0 and 2*pi.

     [g, f] = grpdelay(b,a,n,Fs) evaluates the group delay at n
     frequencies between 0 and Fs/2.

     [g, f] = grpdelay(b,a,n,'whole',Fs) evaluates the group delay at n
     frequencies between 0 and Fs.

     [g, w] = grpdelay(b,a,w) evaluates the group delay at frequencies w
     (radians per sample).

     [g, f] = grpdelay(b,a,f,Fs) evaluates the group delay at
     frequencies f (in Hz).

     grpdelay(...)  plots the group delay vs.  frequency.

     If the denominator of the computation becomes too small, the group
     delay is set to zero.  (The group delay approaches infinity when
     there are poles or zeros very close to the unit circle in the z
     plane.)

     Theory: group delay, g(w) = -d/dw [arg{H(e^jw)}], is the rate of
     change of phase with respect to frequency.  It can be computed as:

                        d/dw H(e^-jw)
                 g(w) = -------------
                          H(e^-jw)

     where

                  H(z) = B(z)/A(z) = sum(b_k z^k)/sum(a_k z^k).

     By the quotient rule,

                             A(z) d/dw B(z) - B(z) d/dw A(z)
                 d/dw H(z) = -------------------------------
                                        A(z) A(z)

     Substituting into the expression above yields:

                         A dB - B dA
                 g(w) =  ----------- = dB/B - dA/A
                             A B

     Note that,

                 d/dw B(e^-jw) = sum(k b_k e^-jwk)
                 d/dw A(e^-jw) = sum(k a_k e^-jwk)

     which is just the FFT of the coefficients multiplied by a ramp.

     As a further optimization when nfft>>length(a), the IIR filter
     (b,a) is converted to the FIR filter conv(b,fliplr(conj(a))).  For
     further details, see
     http://ccrma.stanford.edu/~jos/filters/Numerical_Computation_Group_Delay.html


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 36
Compute the group delay of a filter.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
hann


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 591
 -- Function File: hann (M)
 -- Function File: hann (M, "periodic")
 -- Function File: hann (M, "symmetric")
     Return the filter coefficients of a Hanning window of length M.

     If the optional argument '"periodic"' is given, the periodic form
     of the window is returned.  This is equivalent to the window of
     length M+1 with the last coefficient removed.  The optional
     argument '"symmetric"' is equivalent to not specifying a second
     argument.

     This function exists for MATLAB compatibility only, and is
     equivalent to 'hanning (M)'.

     See also: hanning.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 63
Return the filter coefficients of a Hanning window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
hilbert


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 663
 -- Function File: H = hilbert (F, N, DIM)
     Analytic extension of real valued signal.

     'H = hilbert (F)' computes the extension of the real valued signal
     F to an analytic signal.  If F is a matrix, the transformation is
     applied to each column.  For N-D arrays, the transformation is
     applied to the first non-singleton dimension.

     'real (H)' contains the original signal F.  'imag (H)' contains the
     Hilbert transform of F.

     'hilbert (F, N)' does the same using a length N Hilbert transform.
     The result will also have length N.

     'hilbert (F, [], DIM)' or 'hilbert (F, N, DIM)' does the same along
     dimension DIM.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 41
Analytic extension of real valued signal.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
idct


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 674
 -- Function File: Y = idct (X)
 -- Function File: Y = idct (X, N)
     Compute the inverse discrete cosine transform of X.  If N is given,
     then X is padded or trimmed to length N before computing the
     transform.  If X is a matrix, compute the transform along the
     columns of the the matrix.  The transform is faster if X is
     real-valued and even length.

     The inverse discrete cosine transform X can be defined as follows:

                   N-1
            x[n] = sum w(k) X[k] cos (pi (2n+1) k / 2N ),  n = 0, ..., N-1
                   k=0

     with w(0) = sqrt(1/N) and w(k) = sqrt(2/N), k = 1, ..., N-1

     See also: dct, dct2, idct2, dctmtx.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 51
Compute the inverse discrete cosine transform of X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
idct2


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 287
 -- Function File: Y = idct2 (X)
 -- Function File: Y = idct2 (X, M, N)
 -- Function File: Y = idct2 (X, [M, N])
     Compute the inverse 2-D discrete cosine transform of matrix X.  If
     M and N are specified, the input is either padded or truncated to
     have M rows and N columns.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 62
Compute the inverse 2-D discrete cosine transform of matrix X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
idst


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 331
 -- Function File: Y = idst (X)
 -- Function File: Y = idst (X, N)
     Computes the inverse type I discrete sine transform of Y.  If N is
     given, then Y is padded or trimmed to length N before computing the
     transform.  If Y is a matrix, compute the transform along the
     columns of the the matrix.

     See also: dst.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 57
Computes the inverse type I discrete sine transform of Y.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
ifht


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 741
 -- Function File: M = ifht (D, N, DIM)
     Calculate the inverse Fast Hartley Transform of real input D.  If D
     is a matrix, the inverse Hartley transform is calculated along the
     columns by default.  The options N and DIM are similar to the
     options of FFT function.

     The forward and inverse Hartley transforms are the same (except for
     a scale factor of 1/N for the inverse hartley transform), but
     implemented using different functions.

     The definition of the forward hartley transform for vector d, m[K]
     = 1/N \sum_{i=0}^{N-1} d[i]*(cos[K*2*pi*i/N] + sin[K*2*pi*i/N]),
     for 0 <= K < N. m[K] = 1/N \sum_{i=0}^{N-1} d[i]*CAS[K*i], for 0 <=
     K < N.

          ifht(1:4)

     See also: fht, fft.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 61
Calculate the inverse Fast Hartley Transform of real input D.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
ifwht


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1006
 -- Function File: ifwht (X)
 -- Function File: ifwht (X, N)
 -- Function File: ifwht (X, N, ORDER)
     Compute the inverse Walsh-Hadamard transform of X using the Fast
     Walsh-Hadamard Transform (FWHT) algorithm.  If the input is a
     matrix, the inverse FWHT is calculated along the columns of X.

     The number of elements of X must be a power of 2; if not, the input
     will be extended and filled with zeros.  If a second argument is
     given, the input is truncated or extended to have length N.

     The third argument specifies the ORDER in which the returned
     inverse Walsh-Hadamard transform coefficients should be arranged.
     The ORDER may be any of the following strings:

     "sequency"
          The coefficients are returned in sequency order.  This is the
          default if ORDER is not given.

     "hadamard"
          The coefficients are returned in Hadamard order.

     "dyadic"
          The coefficients are returned in Gray code order.

     See also: fwht.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Compute the inverse Walsh-Hadamard transform of X using the Fast
Walsh-Hadamard 



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
iirlp2mb


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1305
 IIR Low Pass Filter to Multiband Filter Transformation

 [Num,Den,AllpassNum,AllpassDen] = iirlp2mb(B,A,Wo,Wt)
 [Num,Den,AllpassNum,AllpassDen] = iirlp2mb(B,A,Wo,Wt,Pass)

 Num,Den:               numerator,denominator of the transformed filter
 AllpassNum,AllpassDen: numerator,denominator of allpass transform,
 B,A:                   numerator,denominator of prototype low pass filter
 Wo:                    normalized_angular_frequency/pi to be transformed
 Wt:                    [phi=normalized_angular_frequencies]/pi target vector
 Pass:                  This parameter may have values 'pass' or 'stop'.  If
                        not given, it defaults to the value of 'pass'.

 With normalized ang. freq. targets 0 < phi(1) <  ... < phi(n) < pi radians

 for Pass == 'pass', the target multiband magnitude will be:
       --------       ----------        -----------...
      /        \     /          \      /            .
 0   phi(1) phi(2)  phi(3)   phi(4)   phi(5)   (phi(6))    pi

 for Pass == 'stop', the target multiband magnitude will be:
 -------      ---------        ----------...
        \    /         \      /           .
 0   phi(1) phi(2)  phi(3)   phi(4)  (phi(5))              pi

 Example of use:
 [B, A] = butter(6, 0.5);
 [Num, Den] = iirlp2mb(B, A, 0.5, [.2 .4 .6 .8]);



# name: <cell-element>
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# length: 55
 IIR Low Pass Filter to Multiband Filter Transformation



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
impinvar


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 869
 -- Function File: [B_OUT, A_OUT] = impinvar (B, A, FS, TOL)
 -- Function File: [B_OUT, A_OUT] = impinvar (B, A, FS)
 -- Function File: [B_OUT, A_OUT] = impinvar (B, A)
     Converts analog filter with coefficients B and A to digital,
     conserving impulse response.

     If FS is not specified, or is an empty vector, it defaults to 1Hz.

     If TOL is not specified, it defaults to 0.0001 (0.1%) This function
     does the inverse of impinvar so that the following example should
     restore the original values of A and B.

     'invimpinvar' implements the reverse of this function.
          [b, a] = impinvar (b, a);
          [b, a] = invimpinvar (b, a);

     Reference: Thomas J. Cavicchi (1996) "Impulse invariance and
     multiple-order poles".  IEEE transactions on signal processing, Vol
     44 (9): 2344-2347

     See also: bilinear, invimpinvar.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Converts analog filter with coefficients B and A to digital, conserving
impulse 



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
impz


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 744
 -- Function File: [X, T] = impz (B)
 -- Function File: [X, T] = impz (B, A)
 -- Function File: [X, T] = impz (B, A, N)
 -- Function File: [X, T] = impz (B, A, N, FS)
 -- Function File: impz (...)

     Generate impulse-response characteristics of the filter.  The
     filter coefficients correspond to the the z-plane rational function
     with numerator b and denominator a.  If a is not specified, it
     defaults to 1.  If n is not specified, or specified as [], it will
     be chosen such that the signal has a chance to die down to -120dB,
     or to not explode beyond 120dB, or to show five periods if there is
     no significant damping.  If no return arguments are requested, plot
     the results.

     See also: freqz, zplane.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 56
Generate impulse-response characteristics of the filter.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
interp


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 733
 -- Function File: Y = interp (X, Q)
 -- Function File: Y = interp (X, Q, N)
 -- Function File: Y = interp (X, Q, N, WC)

     Upsample the signal x by a factor of q, using an order 2*q*n+1 FIR
     filter.  Note that q must be an integer for this rate change
     method.  n defaults to 4 and Wc defaults to 0.5.

     Example
          # Generate a signal.
          t=0:0.01:2; x=chirp(t,2,.5,10,'quadratic')+sin(2*pi*t*0.4);
          y = interp(x(1:4:length(x)),4,4,1);   # interpolate a sub-sample
          stem(t(1:121)*1000,x(1:121),"-g;Original;"); hold on;
          stem(t(1:121)*1000,y(1:121),"-r;Interpolated;");
          stem(t(1:4:121)*1000,x(1:4:121),"-b;Subsampled;"); hold off;

     See also: decimate, resample.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 74
Upsample the signal x by a factor of q, using an order 2*q*n+1 FIR
filter.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
invfreq


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1689
 usage: [B,A] = invfreq(H,F,nB,nA)
        [B,A] = invfreq(H,F,nB,nA,W)
        [B,A] = invfreq(H,F,nB,nA,W,[],[],plane)
        [B,A] = invfreq(H,F,nB,nA,W,iter,tol,plane)

 Fit filter B(z)/A(z) or B(s)/A(s) to complex frequency response at
 frequency points F. A and B are real polynomial coefficients of order
 nA and nB respectively.  Optionally, the fit-errors can be weighted vs
 frequency according to the weights W. Also, the transform plane can be
 specified as either 's' for continuous time or 'z' for discrete time. 'z'
 is chosen by default.  Eventually, Steiglitz-McBride iterations will be
 specified by iter and tol.

 H: desired complex frequency response
     It is assumed that A and B are real polynomials, hence H is one-sided.
 F: vector of frequency samples in radians
 nA: order of denominator polynomial A
 nB: order of numerator polynomial B
 plane='z': F on unit circle (discrete-time spectra, z-plane design)
 plane='s': F on jw axis     (continuous-time spectra, s-plane design)
 H(k) = spectral samples of filter frequency response at points zk,
  where zk=exp(sqrt(-1)*F(k)) when plane='z' (F(k) in [0,.5])
     and zk=(sqrt(-1)*F(k)) when plane='s' (F(k) nonnegative)
 Example:
     [B,A] = butter(12,1/4);
     [H,w] = freqz(B,A,128);
     [Bh,Ah] = invfreq(H,F,4,4);
     Hh = freqz(Bh,Ah);
     disp(sprintf('||frequency response error|| = %f',norm(H-Hh)));

 References: J. O. Smith, "Techniques for Digital Filter Design and System
      Identification with Application to the Violin, Ph.D. Dissertation,
      Elec. Eng. Dept., Stanford University, June 1983, page 50; or,

 http://ccrma.stanford.edu/~jos/filters/FFT_Based_Equation_Error_Method.html



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
 usage: [B,A] = invfreq(H,F,nB,nA)
        [B,A] = invfreq(H,F,nB,nA,W)
        



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
invfreqs


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 943
 Usage: [B,A] = invfreqs(H,F,nB,nA)
        [B,A] = invfreqs(H,F,nB,nA,W)
        [B,A] = invfreqs(H,F,nB,nA,W,iter,tol,'trace')

 Fit filter B(s)/A(s)to the complex frequency response H at frequency
 points F.  A and B are real polynomial coefficients of order nA and nB.
 Optionally, the fit-errors can be weighted vs frequency according to
 the weights W.
 Note: all the guts are in invfreq.m

 H: desired complex frequency response
 F: frequency (must be same length as H)
 nA: order of the denominator polynomial A
 nB: order of the numerator polynomial B
 W: vector of weights (must be same length as F)

 Example:
       B = [1/2 1];
       A = [1 1];
       w = linspace(0,4,128);
       H = freqs(B,A,w);
       [Bh,Ah] = invfreqs(H,w,1,1);
       Hh = freqs(Bh,Ah,w);
       plot(w,[abs(H);abs(Hh)])
       legend('Original','Measured');
       err = norm(H-Hh);
       disp(sprintf('L2 norm of frequency response error = %f',err));



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
 Usage: [B,A] = invfreqs(H,F,nB,nA)
        [B,A] = invfreqs(H,F,nB,nA,W)
      



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
invfreqz


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 820
 usage: [B,A] = invfreqz(H,F,nB,nA)
        [B,A] = invfreqz(H,F,nB,nA,W)
        [B,A] = invfreqz(H,F,nB,nA,W,iter,tol,'trace')

 Fit filter B(z)/A(z)to the complex frequency response H at frequency
 points F.  A and B are real polynomial coefficients of order nA and nB.
 Optionally, the fit-errors can be weighted vs frequency according to
 the weights W.
 Note: all the guts are in invfreq.m

 H: desired complex frequency response
 F: normalized frequency (0 to pi) (must be same length as H)
 nA: order of the denominator polynomial A
 nB: order of the numerator polynomial B
 W: vector of weights (must be same length as F)

 Example:
     [B,A] = butter(4,1/4);
     [H,F] = freqz(B,A);
     [Bh,Ah] = invfreq(H,F,4,4);
     Hh = freqz(Bh,Ah);
     disp(sprintf('||frequency response error|| = %f',norm(H-Hh)));



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
 usage: [B,A] = invfreqz(H,F,nB,nA)
        [B,A] = invfreqz(H,F,nB,nA,W)
      



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 11
invimpinvar


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 815
 -- Function File: [B_OUT, A_OUT] = invimpinvar (B, A, FS, TOL)
 -- Function File: [B_OUT, A_OUT] = invimpinvar (B, A, FS)
 -- Function File: [B_OUT, A_OUT] = invimpinvar (B, A)
     Converts digital filter with coefficients B and A to analog,
     conserving impulse response.

     This function does the inverse of impinvar so that the following
     example should restore the original values of A and B.
          [b, a] = impinvar (b, a);
          [b, a] = invimpinvar (b, a);

     If FS is not specified, or is an empty vector, it defaults to 1Hz.

     If TOL is not specified, it defaults to 0.0001 (0.1%)

     Reference: Thomas J. Cavicchi (1996) "Impulse invariance and
     multiple-order poles".  IEEE transactions on signal processing, Vol
     40 (9): 2344-2347

     See also: bilinear, impinvar.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Converts digital filter with coefficients B and A to analog, conserving
impulse 



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
kaiser


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 637
 -- Function File: kaiser (M)
 -- Function File: kaiser (M, BETA)

     Return the filter coefficients of a Kaiser window of length M.  The
     Fourier transform of the window has a stop-band attenuation that is
     derived from the parameter BETA.

     For the definition of the Kaiser window, see A. V. Oppenheim & R.
     W. Schafer, "Discrete-Time Signal Processing".

     The continuous version of width m centered about x=0 is:

                  besseli(0, beta * sqrt(1-(2*x/m).^2))
          k(x) =  -------------------------------------,  m/2 <= x <= m/2
                         besseli(0, beta)

     See also: kaiserord.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 62
Return the filter coefficients of a Kaiser window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 9
kaiserord


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1723
 -- Function File: [N, WN, BETA, FTYPE] = kaiserord (F, M, DEV)
 -- Function File: [...] = kaiserord (F, M, DEV, FS)

     Return the parameters needed to produce a filter of the desired
     specification from a Kaiser window.  The vector F contains pairs of
     frequency band edges in the range [0,1].  The vector M specifies
     the magnitude response for each band.  The values of M must be zero
     for all stop bands and must have the same magnitude for all pass
     bands.  The deviation of the filter DEV can be specified as a
     scalar or a vector of the same length as M.  The optional sampling
     rate FS can be used to indicate that F is in Hz in the range
     [0,FS/2].

     The returned value N is the required order of the filter (the
     length of the filter minus 1).  The vector WN contains the band
     edges of the filter suitable for passing to 'fir1'.  The value BETA
     is the parameter of the Kaiser window of length N+1 to shape the
     filter.  The string FTYPE contains the type of filter to specify to
     'fir1'.

     The Kaiser window parameters n and beta are computed from the
     relation between ripple (A=-20*log10(dev)) and transition width (dw
     in radians) discovered empirically by Kaiser:

                    / 0.1102(A-8.7)                        A > 50
             beta = | 0.5842(A-21)^0.4 + 0.07886(A-21)     21 <= A <= 50
                    \ 0.0                                  A < 21

             n = (A-8)/(2.285 dw)

     Example:
          [n, w, beta, ftype] = kaiserord ([1000, 1200], [1, 0], [0.05, 0.05], 11025);
          b = fir1 (n, w, kaiser (n+1, beta), ftype, "noscale");
          freqz (b, 1, [], 11025);

     See also: fir1, kaiser.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Return the parameters needed to produce a filter of the desired
specification fr



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
levinson


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 892
 -- Function File: [A, V, REF] = levinson (ACF)
 -- Function File: [...] = levinson (ACF, P)

     Use the Durbin-Levinson algorithm to solve: toeplitz(acf(1:p)) * x
     = -acf(2:p+1).  The solution [1, x'] is the denominator of an all
     pole filter approximation to the signal x which generated the
     autocorrelation function acf.

     acf is the autocorrelation function for lags 0 to p.  p defaults to
     length(acf)-1.  Returns a=[1, x'] the denominator filter
     coefficients.  v= variance of the white noise = square of the
     numerator constant ref = reflection coefficients = coefficients of
     the lattice implementation of the filter Use freqz(sqrt(v),a) to
     plot the power spectrum.

     REFERENCE [1] Steven M. Kay and Stanley Lawrence Marple Jr.:
     "Spectrum analysis - a modern perspective", Proceedings of the
     IEEE, Vol 69, pp 1380-1419, Nov., 1981


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Use the Durbin-Levinson algorithm to solve: toeplitz(acf(1:p)) * x =
-acf(2:p+1)



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
marcumq


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 958
 -- Function File: Q = marcumq (A, B)
 -- Function File: Q = marcumq (A, B, M)
 -- Function File: Q = marcumq (A, B, M, TOL)

     Compute the generalized Marcum Q function of order M with
     noncentrality parameter A and argument B.  If the order M is
     omitted it defaults to 1.  An optional relative tolerance TOL may
     be included, the default is 'eps'.

     If the input arguments are commensurate vectors, this function will
     produce a table of values.

     This function computes Marcum's Q function using the infinite
     Bessel series, truncated when the relative error is less than the
     specified tolerance.  The accuracy is limited by that of the Bessel
     functions, so reducing the tolerance is probably not useful.

     Reference: Marcum, "Tables of Q Functions", Rand Corporation.

     Reference: R.T. Short, "Computation of Noncentral Chi-squared and
     Rice Random Variables", www.phaselockedsystems.com/publications


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Compute the generalized Marcum Q function of order M with noncentrality
paramete



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
mexihat


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 87
 -- Function File: [PSI, X] = mexihat (LB, UB, N)
     Compute the Mexican hat wavelet.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 32
Compute the Mexican hat wavelet.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
meyeraux


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 86
 -- Function File: Y = meyeraux (X)
     Compute the Meyer wavelet auxiliary function.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 45
Compute the Meyer wavelet auxiliary function.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
morlet


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 81
 -- Function File: [PSI, X] = morlet (LB, UB, N)
     Compute the Morlet wavelet.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 27
Compute the Morlet wavelet.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 9
movingrms


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 392
 -- Function File: [RMSX,W] = movingrms (X,W,RC,FS=1)
     Calculate moving RMS value of the signal in X.

     The signal is convoluted against a sigmoid window of width W and
     risetime RC.  The units of these parameters are relative to the
     value of the sampling frequency given in FS (Default value = 1).

     Run 'demo movingrms' to see an example.

     See also: sigmoid_train.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 46
Calculate moving RMS value of the signal in X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
mscohere


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 550
 -- Function File: [PXX, FREQ] = mscohere (X, Y)
 -- Function File: [...] = mscohere (X, Y, WINDOW)
 -- Function File: [...] = mscohere (X, Y, WINDOW, OVERLAP)
 -- Function File: [...] = mscohere (X, Y, WINDOW, OVERLAP, NFFT)
 -- Function File: [...] = mscohere (X, Y, WINDOW, OVERLAP, NFFT, FS)
 -- Function File: [...] = mscohere (X, Y, WINDOW, OVERLAP, NFFT, FS,
          RANGE)
 -- Function File: mscohere (...)

     Estimate (mean square) coherence of signals X and Y.  Use the Welch
     (1967) periodogram/FFT method.

     See also: pwelch.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 52
Estimate (mean square) coherence of signals X and Y.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
ncauer


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 380
 usage: [Zz, Zp, Zg] = ncauer(Rp, Rs, n)

 Analog prototype for Cauer filter.
 [z, p, g]=ncauer(Rp, Rs, ws)
 Rp = Passband ripple
 Rs = Stopband ripple
 Ws = Desired order

 References:

 - Serra, Celso Penteado, Teoria e Projeto de Filtros, Campinas: CARTGRAF,
   1983.
 - Lamar, Marcus Vinicius, Notas de aula da disciplina TE 456 - Circuitos
   Analogicos II, UFPR, 2001/2002.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 40
 usage: [Zz, Zp, Zg] = ncauer(Rp, Rs, n)



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 10
nuttallwin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 559
 -- Function File: nuttallwin (M)
 -- Function File: nuttallwin (M, "periodic")
 -- Function File: nuttallwin (M, "symmetric")
     Return the filter coefficients of a Blackman-Harris window defined
     by Nuttall of length M.

     If the optional argument '"periodic"' is given, the periodic form
     of the window is returned.  This is equivalent to the window of
     length M+1 with the last coefficient removed.  The optional
     argument '"symmetric"' is equivalent to not specifying a second
     argument.

     See also: blackman, blackmanharris.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Return the filter coefficients of a Blackman-Harris window defined by
Nuttall of



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 9
parzenwin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 135
 -- Function File: parzenwin (M)
     Return the filter coefficients of a Parzen window of length M.

     See also: rectwin, bartlett.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 62
Return the filter coefficients of a Parzen window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
pburg


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3777
 usage:
    [psd,f_out] = pburg(x,poles,freq,Fs,range,method,plot_type,criterion)

 Calculate Burg maximum-entropy power spectral density.
 The functions "arburg" and "ar_psd" do all the work.
 See "help arburg" and "help ar_psd" for further details.

 ARGUMENTS:
     All but the first two arguments are optional and may be empty.
   x       %% [vector] sampled data

   poles   %% [integer scalar] required number of poles of the AR model

   freq    %% [real vector] frequencies at which power spectral density
           %%               is calculated
           %% [integer scalar] number of uniformly distributed frequency
           %%          values at which spectral density is calculated.
           %%          [default=256]

   Fs      %% [real scalar] sampling frequency (Hertz) [default=1]


 CONTROL-STRING ARGUMENTS -- each of these arguments is a character string.
   Control-string arguments can be in any order after the other arguments.


   range   %% 'half',  'onesided' : frequency range of the spectrum is
           %%       from zero up to but not including sample_f/2.  Power
           %%       from negative frequencies is added to the positive
           %%       side of the spectrum.
           %% 'whole', 'twosided' : frequency range of the spectrum is
           %%       -sample_f/2 to sample_f/2, with negative frequencies
           %%       stored in "wrap around" order after the positive
           %%       frequencies; e.g. frequencies for a 10-point 'twosided'
           %%       spectrum are 0 0.1 0.2 0.3 0.4 0.5 -0.4 -0.3 -0.2 -0.1
           %% 'shift', 'centerdc' : same as 'whole' but with the first half
           %%       of the spectrum swapped with second half to put the
           %%       zero-frequency value in the middle. (See "help
           %%       fftshift". If "freq" is vector, 'shift' is ignored.
           %% If model coefficients "ar_coeffs" are real, the default
           %% range is 'half', otherwise default range is 'whole'.

   method  %% 'fft':  use FFT to calculate power spectral density.
           %% 'poly': calculate spectral density as a polynomial of 1/z
           %% N.B. this argument is ignored if the "freq" argument is a
           %%      vector.  The default is 'poly' unless the "freq"
           %%      argument is an integer power of 2.

 plot_type %% 'plot', 'semilogx', 'semilogy', 'loglog', 'squared' or 'db':
           %% specifies the type of plot.  The default is 'plot', which
           %% means linear-linear axes. 'squared' is the same as 'plot'.
           %% 'dB' plots "10*log10(psd)".  This argument is ignored and a
           %% spectrum is not plotted if the caller requires a returned
           %% value.

 criterion %% [optional string arg]  model-selection criterion.  Limits
           %%       the number of poles so that spurious poles are not
           %%       added when the whitened data has no more information
           %%       in it (see Kay & Marple, 1981). Recognized values are
           %%  'AKICc' -- approximate corrected Kullback information
           %%             criterion (recommended),
           %%   'KIC'  -- Kullback information criterion
           %%   'AICc' -- corrected Akaike information criterion
           %%   'AIC'  -- Akaike information criterion
           %%   'FPE'  -- final prediction error" criterion
           %% The default is to NOT use a model-selection criterion

 RETURNED VALUES:
     If return values are not required by the caller, the spectrum
     is plotted and nothing is returned.
   psd       %% [real vector] power-spectral density estimate
   f_out     %% [real vector] frequency values

 HINTS
   This function is a wrapper for arburg and ar_psd.
   See "help arburg", "help ar_psd".



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
 usage:
    [psd,f_out] = pburg(x,poles,freq,Fs,range,method,plot_type,criterion



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 15
pei_tseng_notch


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 691
 -- Function File: [B, A] = pei_tseng_notch (FREQUENCIES, BANDWIDTHS)
     Return coefficients for an IIR notch-filter with one or more filter
     frequencies and according (very narrow) bandwidths to be used with
     'filter' or 'filtfilt'.  The filter construction is based on an
     allpass which performs a reversal of phase at the filter
     frequencies.  Thus, the mean of the phase-distorted and the
     original signal has the respective frequencies removed.  See the
     demo for an illustration.

     Original source: Pei, Soo-Chang, and Chien-Cheng Tseng "IIR
     Multiple Notch Filter Design Based on Allpass Filter" 1996 IEEE
     Tencon doi: 10.1109/TENCON.1996.608814)


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Return coefficients for an IIR notch-filter with one or more filter
frequencies 



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
polystab


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 156
 b = polystab(a)

 Stabilize the polynomial transfer function by replacing all roots
 outside the unit circle with their reflection inside the unit circle.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 16
 b = polystab(a)



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 9
primitive


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 743
 -- Function File: F = primitive (F, T, F0)
     Calculates the primitive of a function.

     The function approximates the primitive (indefinite integral) of
     the univariate function handle F with constant of integration F0.
     The output is the primitive evaluated at the points T.  The vector
     T must be ordered and ascending.

     This function is a fancy way of calculating the cumulative sum,

     'F = cumsum (f(t).*diff (t)([1 1:end]))'.

     Example:
          f = @(t) sin(2*pi*3*t);
          t = [0; sort(rand(100,1))];
          F = primitive (f,t,0);
          t_true = linspace(0,1,1e3).';
          F_true = (1 - cos(2*pi*3*t_true))/(2*pi*3);
          plot (t,F,'.',t_true,F_true)

     See also: quadgk, cumsum.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 39
Calculates the primitive of a function.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
pulstran


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1434
 -- Function File: Y = pulstran (T, D, FUNC, ...)
 -- Function File: Y = pulstran (T, D, P)
 -- Function File: Y = pulstran (T, D, P, FS)
 -- Function File: Y = pulstran (T, D, P, FS, METHOD)

     Generate the signal y=sum(func(t+d,...))  for each d.  If d is a
     matrix of two columns, the first column is the delay d and the
     second column is the amplitude a, and y=sum(a*func(t+d)) for each
     d,a.  Clearly, func must be a function which accepts a vector of
     times.  Any extra arguments needed for the function must be tagged
     on the end.

     Example:
          fs = 11025;  # arbitrary sample rate
          f0 = 100;    # pulse train sample rate
          w = 0.001;   # pulse width of 1 millisecond
          auplot (pulstran (0:1/fs:0.1, 0:1/f0:0.1, "rectpuls", w), fs);

     If instead of a function name you supply a pulse shape sampled at
     frequency Fs (default 1 Hz), an interpolated version of the pulse
     is added at each delay d.  The interpolation stays within the the
     time range of the delayed pulse.  The interpolation method defaults
     to linear, but it can be any interpolation method accepted by the
     function interp1.

     Example:
          fs = 11025;      # arbitrary sample rate
          f0 = 100;        # pulse train sample rate
          w = boxcar(10);  # pulse width of 1 millisecond at 10 kHz
          auplot (pulstran (0:1/fs:0.1, 0:1/f0:0.1, w, 10000), fs);


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 53
Generate the signal y=sum(func(t+d,...))  for each d.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
pwelch


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7135
 USAGE:
   [spectra,freq] = pwelch(x,window,overlap,Nfft,Fs,
                           range,plot_type,detrend,sloppy)
     Estimate power spectral density of data "x" by the Welch (1967)
     periodogram/FFT method.  All arguments except "x" are optional.
         The data is divided into segments.  If "window" is a vector, each
     segment has the same length as "window" and is multiplied by "window"
     before (optional) zero-padding and calculation of its periodogram. If
     "window" is a scalar, each segment has a length of "window" and a
     Hamming window is used.
         The spectral density is the mean of the periodograms, scaled so that
     area under the spectrum is the same as the mean square of the
     data.  This equivalence is supposed to be exact, but in practice there
     is a mismatch of up to 0.5% when comparing area under a periodogram
     with the mean square of the data.

  [spectra,freq] = pwelch(x,y,window,overlap,Nfft,Fs,
                          range,plot_type,detrend,sloppy,results)
     Two-channel spectrum analyser.  Estimate power spectral density, cross-
     spectral density, transfer function and/or coherence functions of time-
     series input data "x" and output data "y" by the Welch (1967)
     periodogram/FFT method.
       pwelch treats the second argument as "y" if there is a control-string
     argument "cross", "trans", "coher" or "ypower"; "power" does not force
     the 2nd argument to be treated as "y".  All other arguments are
     optional.  All spectra are returned in matrix "spectra".

  [spectra,Pxx_ci,freq] = pwelch(x,window,overlap,Nfft,Fs,conf,
                                 range,plot_type,detrend,sloppy)
  [spectra,Pxx_ci,freq] = pwelch(x,y,window,overlap,Nfft,Fs,conf,
                                 range,plot_type,detrend,sloppy,results)
     Estimates confidence intervals for the spectral density.
     See Hint (7) below for compatibility options.  Confidence level "conf"
     is the 6th or 7th numeric argument.  If "results" control-string
     arguments are used, one of them must be "power" when the "conf"
     argument is present; pwelch can estimate confidence intervals only for
     the power spectrum of the "x" data.  It does not know how to estimate
     confidence intervals of the cross-power spectrum, transfer function or
     coherence; if you can suggest a good method, please send a bug report.

 ARGUMENTS
 All but the first argument are optional and may be empty, except that
 the "results" argument may require the second argument to be "y".

 x           %% [non-empty vector] system-input time-series data
 y           %% [non-empty vector] system-output time-series data

 window      %% [real vector] of window-function values between 0 and 1; the
             %%       data segment has the same length as the window.
             %%       Default window shape is Hamming.
             %% [integer scalar] length of each data segment.  The default
             %%       value is window=sqrt(length(x)) rounded up to the
             %%       nearest integer power of 2; see 'sloppy' argument.

 overlap     %% [real scalar] segment overlap expressed as a multiple of
             %%       window or segment length.   0 <= overlap < 1,
             %%       The default is overlap=0.5 .

 Nfft        %% [integer scalar] Length of FFT.  The default is the length
             %%       of the "window" vector or has the same value as the
             %%       scalar "window" argument.  If Nfft is larger than the
             %%       segment length, "seg_len", the data segment is padded
             %%       with "Nfft-seg_len" zeros.  The default is no padding.
             %%       Nfft values smaller than the length of the data
             %%       segment (or window) are ignored silently.

 Fs          %% [real scalar] sampling frequency (Hertz); default=1.0

 conf        %% [real scalar] confidence level between 0 and 1.  Confidence
             %%       intervals of the spectral density are estimated from
             %%       scatter in the periodograms and are returned as Pxx_ci.
             %%       Pxx_ci(:,1) is the lower bound of the confidence
             %%       interval and Pxx_ci(:,2) is the upper bound.  If there
             %%       are three return values, or conf is an empty matrix,
             %%       confidence intervals are calculated for conf=0.95 .
             %%       If conf is zero or is not given, confidence intervals
             %%       are not calculated. Confidence intervals can be
             %%       obtained only for the power spectral density of x;
             %%       nothing else.

 CONTROL-STRING ARGUMENTS -- each of these arguments is a character string.
   Control-string arguments must be after the other arguments but can be in
   any order.

 range     %% 'half',  'onesided' : frequency range of the spectrum is
           %%       zero up to but not including Fs/2.  Power from
           %%       negative frequencies is added to the positive side of
           %%       the spectrum, but not at zero or Nyquist (Fs/2)
           %%       frequencies.  This keeps power equal in time and
           %%       spectral domains.  See reference [2].
           %% 'whole', 'twosided' : frequency range of the spectrum is
           %%       -Fs/2 to Fs/2, with negative frequencies
           %%       stored in "wrap around" order after the positive
           %%       frequencies; e.g. frequencies for a 10-point 'twosided'
           %%       spectrum are 0 0.1 0.2 0.3 0.4 0.5 -0.4 -0.3 -0.2 -0.1
           %% 'shift', 'centerdc' : same as 'whole' but with the first half
           %%       of the spectrum swapped with second half to put the
           %%       zero-frequency value in the middle. (See "help
           %%       fftshift".
           %% If data (x and y) are real, the default range is 'half',
           %% otherwise default range is 'whole'.

 plot_type %% 'plot', 'semilogx', 'semilogy', 'loglog', 'squared' or 'db':
           %% specifies the type of plot.  The default is 'plot', which
           %% means linear-linear axes. 'squared' is the same as 'plot'.
           %% 'dB' plots "10*log10(psd)".  This argument is ignored and a
           %% spectrum is not plotted if the caller requires a returned
           %% value.

 detrend   %% 'no-strip', 'none' -- do NOT remove mean value from the data
           %% 'short', 'mean' -- remove the mean value of each segment from
           %%                    each segment of the data.
           %% 'linear',       -- remove linear trend from each segment of
           %%                    the data.
           %% 'long-mean'     -- remove the mean value from the data before
           %%              splitting it into segments.  This is the default.

   sloppy  %% 'sloppy': FFT length is rounded up to the nearest integer
           %%       power of 2 by zero padding.  FFT length is adjusted
           %%       after addition of padding by explicit Nfft argument.
           %%       The default is to use exactly the FFT and window/



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
 USAGE:
   [spectra,freq] = pwelch(x,window,overlap,Nfft,Fs,
                   



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
pyulear


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3135
 usage:
    [psd,f_out] = pyulear(x,poles,freq,Fs,range,method,plot_type)

 Calculates a Yule-Walker autoregressive (all-pole) model of the data "x"
 and computes the power spectrum of the model.  This is a wrapper for
 functions "aryule" and "ar_psd" which perform the argument checking.
 See "help aryule" and "help ar_psd" for further details.

 ARGUMENTS:
     All but the first two arguments are optional and may be empty.
   x       %% [vector] sampled data

   poles   %% [integer scalar] required number of poles of the AR model

   freq    %% [real vector] frequencies at which power spectral density
           %%               is calculated
           %% [integer scalar] number of uniformly distributed frequency
           %%          values at which spectral density is calculated.
           %%          [default=256]

   Fs      %% [real scalar] sampling frequency (Hertz) [default=1]


 CONTROL-STRING ARGUMENTS -- each of these arguments is a character string.
   Control-string arguments can be in any order after the other arguments.


   range   %% 'half',  'onesided' : frequency range of the spectrum is
           %%       from zero up to but not including sample_f/2.  Power
           %%       from negative frequencies is added to the positive
           %%       side of the spectrum.
           %% 'whole', 'twosided' : frequency range of the spectrum is
           %%       -sample_f/2 to sample_f/2, with negative frequencies
           %%       stored in "wrap around" order after the positive
           %%       frequencies; e.g. frequencies for a 10-point 'twosided'
           %%       spectrum are 0 0.1 0.2 0.3 0.4 0.5 -0.4 -0.3 -0.2 -0.1
           %% 'shift', 'centerdc' : same as 'whole' but with the first half
           %%       of the spectrum swapped with second half to put the
           %%       zero-frequency value in the middle. (See "help
           %%       fftshift". If "freq" is vector, 'shift' is ignored.
           %% If model coefficients "ar_coeffs" are real, the default
           %% range is 'half', otherwise default range is 'whole'.

   method  %% 'fft':  use FFT to calculate power spectrum.
           %% 'poly': calculate power spectrum as a polynomial of 1/z
           %% N.B. this argument is ignored if the "freq" argument is a
           %%      vector.  The default is 'poly' unless the "freq"
           %%      argument is an integer power of 2.

 plot_type %% 'plot', 'semilogx', 'semilogy', 'loglog', 'squared' or 'db':
           %% specifies the type of plot.  The default is 'plot', which
           %% means linear-linear axes. 'squared' is the same as 'plot'.
           %% 'dB' plots "10*log10(psd)".  This argument is ignored and a
           %% spectrum is not plotted if the caller requires a returned
           %% value.

 RETURNED VALUES:
     If return values are not required by the caller, the spectrum
     is plotted and nothing is returned.
   psd     %% [real vector] power-spectrum estimate
   f_out   %% [real vector] frequency values

 HINTS
   This function is a wrapper for aryule and ar_psd.
   See "help aryule", "help ar_psd".



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 73
 usage:
    [psd,f_out] = pyulear(x,poles,freq,Fs,range,method,plot_type)



# name: <cell-element>
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# length: 9
qp_kaiser


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 691
 -- Function File: qp_kaiser (NB, AT)
 -- Function File: qp_kaiser (NB, AT, LINEAR)

     Computes a finite impulse response (FIR) filter for use with a
     quasi-perfect reconstruction polyphase-network filter bank.  This
     version utilizes a Kaiser window to shape the frequency response of
     the designed filter.  Tha number nb of bands and the desired
     attenuation at in the stop-band are given as parameters.

     The Kaiser window is multiplied by the ideal impulse response
     h(n)=a.sinc(a.n) and converted to its minimum-phase version by
     means of a Hilbert transform.

     By using a third non-null argument, the minimum-phase calculation
     is omitted at all.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Computes a finite impulse response (FIR) filter for use with a
quasi-perfect rec



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
rceps


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 816
 -- Function File: [Y, XM] = rceps (X)
     Produce the cepstrum of the signal x, and if desired, the minimum
     phase reconstruction of the signal x.  If x is a matrix, do so for
     each column of the matrix.

     Example:
          f0 = 70; Fs = 10000;                   # 100 Hz fundamental, 10kHz sampling rate
          a = poly (0.985 * exp (1i*pi*[0.1, -0.1, 0.3, -0.3])); # two formants
          s = 0.005 * randn (1024, 1);           # Noise excitation signal
          s(1:Fs/f0:length(s)) = 1;              # Impulse glottal wave
          x = filter (1, a, s);                  # Speech signal in x
          [y, xm] = rceps (x .* hanning (1024)); # cepstrum and min phase reconstruction

     Reference Programs for digital signal processing.  IEEE Press.  New
     York: John Wiley & Sons.  1979.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Produce the cepstrum of the signal x, and if desired, the minimum phase
reconstr



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
rectpuls


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 528
 -- Function File: Y = rectpuls (T)
 -- Function File: Y = rectpuls (T, W)

     Generate a rectangular pulse over the interval [-W/2,W/2), sampled
     at times T.  This is useful with the function 'pulstran' for
     generating a series pulses.

     Example:
          fs = 11025;  # arbitrary sample rate
          f0 = 100;    # pulse train sample rate
          w = 0.3/f0;  # pulse width 3/10th the distance between pulses
          auplot(pulstran(0:1/fs:4/f0, 0:1/f0:4/f0, 'rectpuls', w), fs);

     See also: pulstran.


# name: <cell-element>
# type: sq_string
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# length: 78
Generate a rectangular pulse over the interval [-W/2,W/2), sampled at
times T.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
rectwin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 145
 -- Function File: rectwin (M)
     Return the filter coefficients of a rectangular window of length M.

     See also: boxcar, hamming, hanning.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 67
Return the filter coefficients of a rectangular window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
resample


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 648
 -- Function File: [Y, H] = resample (X, P, Q)
 -- Function File: Y = resample (X, P, Q, H)
     Change the sample rate of X by a factor of P/Q.  This is performed
     using a polyphase algorithm.  The impulse response H of the
     antialiasing filter is either specified or either designed with a
     Kaiser-windowed sinecard.

     Ref [1] J. G. Proakis and D. G. Manolakis, Digital Signal
     Processing: Principles, Algorithms, and Applications, 4th ed.,
     Prentice Hall, 2007.  Chap.  6

     Ref [2] A. V. Oppenheim, R. W. Schafer and J. R. Buck,
     Discrete-time signal processing, Signal processing series,
     Prentice-Hall, 1999


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 47
Change the sample rate of X by a factor of P/Q.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
residued


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1044
 -- Function File: [R, P, F, M] = residued (B, A)
     Compute the partial fraction expansion (PFE) of filter H(z) =
     B(z)/A(z).  In the usual PFE function 'residuez', the IIR part
     (poles P and residues R) is driven _in parallel_ with the FIR part
     (F).  In this variant, the IIR part is driven by the _output_ of
     the FIR part.  This structure can be more accurate in signal
     modeling applications.

     INPUTS: B and A are vectors specifying the digital filter H(z) =
     B(z)/A(z).  See 'help filter' for documentation of the B and A
     filter coefficients.

     RETURNED:
        * R = column vector containing the filter-pole residues
        * P = column vector containing the filter poles
        * F = row vector containing the FIR part, if any
        * M = column vector of pole multiplicities

     EXAMPLES:
          See test residued verbose to see a number of examples.

     For the theory of operation, see
     'http://ccrma.stanford.edu/~jos/filters/residued.html'

     See also: residue, residued.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 72
Compute the partial fraction expansion (PFE) of filter H(z) = B(z)/A(z).



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
residuez


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 745
 -- Function File: [R, P, F, M] = residuez (B, A)
     Compute the partial fraction expansion of filter H(z) = B(z)/A(z).

     INPUTS: B and A are vectors specifying the digital filter H(z) =
     B(z)/A(z).  See 'help filter' for documentation of the B and A
     filter coefficients.

     RETURNED:
        * R = column vector containing the filter-pole residues
        * P = column vector containing the filter poles
        * F = row vector containing the FIR part, if any
        * M = column vector of pole multiplicities

     EXAMPLES:
          See test residuez verbose to see a number of examples.

     For the theory of operation, see
     'http://ccrma.stanford.edu/~jos/filters/residuez.html'

     See also: residue, residued.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 66
Compute the partial fraction expansion of filter H(z) = B(z)/A(z).



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 18
sampled2continuous


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 424
 -- Function File: XT = sampled2continuous (XN, T, T)

     Calculate the x(t) reconstructed from samples x[n] sampled at a
     rate 1/T samples per unit time.

     t is all the instants of time when you need x(t) from x[n]; this
     time is relative to x[0] and not an absolute time.

     This function can be used to calculate sampling rate effects on
     aliasing, actual signal reconstruction from discrete samples.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Calculate the x(t) reconstructed from samples x[n] sampled at a rate 1/T
samples



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
sawtooth


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 661
 -- Function File: Y = sawtooth (T)
 -- Function File: Y = sawtooth (T, WIDTH)
     Generates a sawtooth wave of period '2 * pi' with limits '+1/-1'
     for the elements of T.

     WIDTH is a real number between '0' and '1' which specifies the
     point between '0' and '2 * pi' where the maximum is.  The function
     increases linearly from '-1' to '1' in '[0, 2 * pi * WIDTH]'
     interval, and decreases linearly from '1' to '-1' in the interval
     '[2 * pi * WIDTH, 2 * pi]'.

     If WIDTH is 0.5, the function generates a standard triangular wave.

     If WIDTH is not specified, it takes a value of 1, which is a
     standard sawtooth function.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Generates a sawtooth wave of period '2 * pi' with limits '+1/-1' for the
element



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
schtrig


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 356
 -- Function File: [RMSX,W] = schtrig (X,LVL,RST=1)
     Implements a multisignal Schmitt trigger with levels LVL.

     The triger works along the first dimension of the array X.  When
     'RST==1' the state of the trigger for all signals is set to the low
     state (i.e.  0).

     Run 'demo schtrig' to see an example.

     See also: clustersegment.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 57
Implements a multisignal Schmitt trigger with levels LVL.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
sftrans


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4017
 -- Function File: [SZ, SP, SG] = sftrans (SZ, SP, SG, W, STOP)

     Transform band edges of a generic lowpass filter (cutoff at W=1)
     represented in splane zero-pole-gain form.  W is the edge of the
     target filter (or edges if band pass or band stop).  Stop is true
     for high pass and band stop filters or false for low pass and band
     pass filters.  Filter edges are specified in radians, from 0 to pi
     (the nyquist frequency).

     Theory: Given a low pass filter represented by poles and zeros in
     the splane, you can convert it to a low pass, high pass, band pass
     or band stop by transforming each of the poles and zeros
     individually.  The following table summarizes the transformation:

          Transform         Zero at x                  Pole at x
          ----------------  -------------------------  ------------------------
          Low Pass          zero: Fc x/C               pole: Fc x/C
          S -> C S/Fc       gain: C/Fc                 gain: Fc/C
          ----------------  -------------------------  ------------------------
          High Pass         zero: Fc C/x               pole: Fc C/x
          S -> C Fc/S       pole: 0                    zero: 0
                            gain: -x                   gain: -1/x
          ----------------  -------------------------  ------------------------
          Band Pass         zero: b +- sqrt(b^2-FhFl)  pole: b +- sqrt(b^2-FhFl)
                 S^2+FhFl   pole: 0                    zero: 0
          S -> C --------   gain: C/(Fh-Fl)            gain: (Fh-Fl)/C
                 S(Fh-Fl)   b=x/C (Fh-Fl)/2            b=x/C (Fh-Fl)/2
          ----------------  -------------------------  ------------------------
          Band Stop         zero: b +- sqrt(b^2-FhFl)  pole: b +- sqrt(b^2-FhFl)
                 S(Fh-Fl)   pole: +-sqrt(-FhFl)        zero: +-sqrt(-FhFl)
          S -> C --------   gain: -x                   gain: -1/x
                 S^2+FhFl   b=C/x (Fh-Fl)/2            b=C/x (Fh-Fl)/2
          ----------------  -------------------------  ------------------------
          Bilinear          zero: (2+xT)/(2-xT)        pole: (2+xT)/(2-xT)
               2 z-1        pole: -1                   zero: -1
          S -> - ---        gain: (2-xT)/T             gain: (2-xT)/T
               T z+1
          ----------------  -------------------------  ------------------------

     where C is the cutoff frequency of the initial lowpass filter, Fc
     is the edge of the target low/high pass filter and [Fl,Fh] are the
     edges of the target band pass/stop filter.  With abundant tedious
     algebra, you can derive the above formulae yourself by substituting
     the transform for S into H(S)=S-x for a zero at x or H(S)=1/(S-x)
     for a pole at x, and converting the result into the form:

             H(S)=g prod(S-Xi)/prod(S-Xj)

     The transforms are from the references.  The actual pole-zero-gain
     changes I derived myself.

     Please note that a pole and a zero at the same place exactly
     cancel.  This is significant for High Pass, Band Pass and Band Stop
     filters which create numerous extra poles and zeros, most of which
     cancel.  Those which do not cancel have a "fill-in" effect,
     extending the shorter of the sets to have the same number of as the
     longer of the sets of poles and zeros (or at least split the
     difference in the case of the band pass filter).  There may be
     other opportunistic cancellations but I will not check for them.

     Also note that any pole on the unit circle or beyond will result in
     an unstable filter.  Because of cancellation, this will only happen
     if the number of poles is smaller than the number of zeros and the
     filter is high pass or band pass.  The analytic design methods all
     yield more poles than zeros, so this will not be a problem.

     References:

     Proakis & Manolakis (1992).  Digital Signal Processing.  New York:
     Macmillan Publishing Company.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Transform band edges of a generic lowpass filter (cutoff at W=1)
represented in 



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
sgolay


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1236
 -- Function File: F = sgolay (P, N)
 -- Function File: F = sgolay (P, N, M)
 -- Function File: F = sgolay (P, N, M, TS)
     Computes the filter coefficients for all Savitzsky-Golay smoothing
     filters of order p for length n (odd).  m can be used in order to
     get directly the mth derivative.  In this case, ts is a scaling
     factor.

     The early rows of F smooth based on future values and later rows
     smooth based on past values, with the middle row using half future
     and half past.  In particular, you can use row i to estimate x(k)
     based on the i-1 preceding values and the n-i following values of x
     values as y(k) = F(i,:) * x(k-i+1:k+n-i).

     Normally, you would apply the first (n-1)/2 rows to the first k
     points of the vector, the last k rows to the last k points of the
     vector and middle row to the remainder, but for example if you were
     running on a realtime system where you wanted to smooth based on
     the all the data collected up to the current time, with a lag of
     five samples, you could apply just the filter on row n-5 to your
     window of length n each time you added a new sample.

     Reference: Numerical recipes in C. p 650

     See also: sgolayfilt.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Computes the filter coefficients for all Savitzsky-Golay smoothing
filters of or



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 10
sgolayfilt


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1214
 -- Function File: Y = sgolayfilt (X)
 -- Function File: Y = sgolayfilt (X, P)
 -- Function File: Y = sgolayfilt (X, P, N)
 -- Function File: Y = sgolayfilt (X, P, N, M)
 -- Function File: Y = sgolayfilt (X, P, N, M, TS)
 -- Function File: Y = sgolayfilt (X, P, N, M, TS)
 -- Function File: Y = sgolayfilt (X, F)
     Smooth the data in x with a Savitsky-Golay smoothing filter of
     polynomial order p and length n, n odd, n > p.  By default, p=3 and
     n=p+2 or n=p+3 if p is even.

     If F is given as a matrix, it is expected to be a filter as
     computed by 'sgolay'.

     These filters are particularly good at preserving lineshape while
     removing high frequency squiggles.  Particularly, compare a 5
     sample averager, an order 5 butterworth lowpass filter (cutoff 1/3)
     and sgolayfilt(x, 3, 5), the best cubic estimated from 5 points:

          [b, a] = butter (5, 1/3);
          x = [zeros(1,15), 10*ones(1,10), zeros(1,15)];
          plot (sgolayfilt (x), "r;sgolayfilt;", ...
                filtfilt (ones (1,5)/5, 1, x), "g;5 sample average;", ...
                filtfilt (b, a, x), "c;order 5 butterworth;", ...
                x, "+b;original data;");

     See also: sgolay.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Smooth the data in x with a Savitsky-Golay smoothing filter of
polynomial order 



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
shanwavf


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 100
 -- Function File: [PSI, X] = shanwavf (LB, UB, N, FB, FC)
     Compute the Complex Shannon wavelet.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 36
Compute the Complex Shannon wavelet.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 13
sigmoid_train


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 556
 -- Function File: Y = sigmoid_train (T, RANGES, RC)

     Evaluate a train of sigmoid functions at T.

     The number and duration of each sigmoid is determined from RANGES.
     Each row of RANGES represents a real interval, e.g.  if sigmoid 'i'
     starts at 't=0.1' and ends at 't=0.5', then 'RANGES(i,:) = [0.1
     0.5]'.  The input RC is an array that defines the rising and
     falling time constants of each sigmoid.  Its size must equal the
     size of RANGES.

     Run 'demo sigmoid_train' to some examples of the use of this
     function.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 43
Evaluate a train of sigmoid functions at T.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
sos2tf


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 909
 -- Function File: [B, A] = sos2tf (SOS)
 -- Function File: [B, A] = sos2tf (SOS, G)
     Convert series second-order sections to direct form H(z) =
     B(z)/A(z).

     INPUTS:

        * SOS = matrix of series second-order sections, one per row:
               SOS = [B1.' A1.'; ...; BN.' AN.']
          where 'B1.' = [b0 b1 b2] and A1.' = [1 a1 a2]' for section 1,
          etc.  The b0 entry must be nonzero for each section.  See
          'filter' for documentation of the second-order direct-form
          filter coefficients Bi and Ai.

        * G is an overall gain factor that effectively scales the output
          B vector (or any one of the input Bi vectors).  If not given
          the gain is assumed to be 1.

     RETURNED: B and A are vectors specifying the digital filter H(z) =
     B(z)/A(z).  See 'filter' for further details.

     See also: tf2sos, zp2sos, sos2pz, zp2tf, tf2zp.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 69
Convert series second-order sections to direct form H(z) = B(z)/A(z).



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
sos2zp


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1145
 -- Function File: [Z, P, K] = sos2zp (SOS)
 -- Function File: [Z, P, K] = sos2zp (SOS, G)
     Convert series second-order sections to zeros, poles, and gains
     (pole residues).

     INPUTS:

        * SOS = matrix of series second-order sections, one per row:
               SOS = [B1.' A1.'; ...; BN.' AN.']
          where 'B1.' = [b0 b1 b2] and A1.' = [1 a1 a2]' for section 1,
          etc.  The b0 entry must be nonzero for each section.  See
          'filter' for documentation of the second-order direct-form
          filter coefficients Bi and Ai.

        * G is an overall gain factor that effectively scales any one of
          the input Bi vectors.  If not given the gain is assumed to be
          1.

     RETURNED:
        * Z = column-vector containing all zeros (roots of B(z))
        * P = column-vector containing all poles (roots of A(z))
        * K = overall gain = B(Inf)

     EXAMPLE:
          [z, p, k] = sos2zp ([1 0 1, 1 0 -0.81; 1 0 0, 1 0 0.49])
              => z = [i; -i; 0; 0]
              => p = [0.9, -0.9, 0.7i, -0.7i]
              => k = 1

     See also: zp2sos, sos2tf, tf2sos, zp2tf, tf2zp.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Convert series second-order sections to zeros, poles, and gains (pole
residues).



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
specgram


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5195
 -- Function File: specgram (X)
 -- Function File: specgram (X, N)
 -- Function File: specgram (X, N, FS)
 -- Function File: specgram (X, N, FS, WINDOW)
 -- Function File: specgram (X, N, FS, WINDOW, OVERLAP)
 -- Function File: [S, F, T] = specgram (...)

     Generate a spectrogram for the signal X.  The signal is chopped
     into overlapping segments of length N, and each segment is windowed
     and transformed into the frequency domain using the FFT. The
     default segment size is 256.  If FS is given, it specifies the
     sampling rate of the input signal.  The argument WINDOW specifies
     an alternate window to apply rather than the default of 'hanning
     (N)'.  The argument OVERLAP specifies the number of samples overlap
     between successive segments of the input signal.  The default
     overlap is 'length (WINDOW)/2'.

     If no output arguments are given, the spectrogram is displayed.
     Otherwise, S is the complex output of the FFT, one row per slice, F
     is the frequency indices corresponding to the rows of S, and T is
     the time indices corresponding to the columns of S.

     Example:
             x = chirp([0:0.001:2],0,2,500);  # freq. sweep from 0-500 over 2 sec.
             Fs=1000;                  # sampled every 0.001 sec so rate is 1 kHz
             step=ceil(20*Fs/1000);    # one spectral slice every 20 ms
             window=ceil(100*Fs/1000); # 100 ms data window
             specgram(x, 2^nextpow2(window), Fs, window, window-step);

             ## Speech spectrogram
             [x, Fs] = auload(file_in_loadpath("sample.wav")); # audio file
             step = fix(5*Fs/1000);     # one spectral slice every 5 ms
             window = fix(40*Fs/1000);  # 40 ms data window
             fftn = 2^nextpow2(window); # next highest power of 2
             [S, f, t] = specgram(x, fftn, Fs, window, window-step);
             S = abs(S(2:fftn*4000/Fs,:)); # magnitude in range 0<f<=4000 Hz.
             S = S/max(S(:));           # normalize magnitude so that max is 0 dB.
             S = max(S, 10^(-40/10));   # clip below -40 dB.
             S = min(S, 10^(-3/10));    # clip above -3 dB.
             imagesc (t, f, log(S));    # display in log scale
             set (gca, "ydir", "normal"); # put the 'y' direction in the correct direction

     The choice of window defines the time-frequency resolution.  In
     speech for example, a wide window shows more harmonic detail while
     a narrow window averages over the harmonic detail and shows more
     formant structure.  The shape of the window is not so critical so
     long as it goes gradually to zero on the ends.

     Step size (which is window length minus overlap) controls the
     horizontal scale of the spectrogram.  Decrease it to stretch, or
     increase it to compress.  Increasing step size will reduce time
     resolution, but decreasing it will not improve it much beyond the
     limits imposed by the window size (you do gain a little bit,
     depending on the shape of your window, as the peak of the window
     slides over peaks in the signal energy).  The range 1-5 msec is
     good for speech.

     FFT length controls the vertical scale.  Selecting an FFT length
     greater than the window length does not add any information to the
     spectrum, but it is a good way to interpolate between frequency
     points which can make for prettier spectrograms.

     After you have generated the spectral slices, there are a number of
     decisions for displaying them.  First the phase information is
     discarded and the energy normalized:

     S = abs(S); S = S/max(S(:));

     Then the dynamic range of the signal is chosen.  Since information
     in speech is well above the noise floor, it makes sense to
     eliminate any dynamic range at the bottom end.  This is done by
     taking the max of the magnitude and some minimum energy such as
     minE=-40dB. Similarly, there is not much information in the very
     top of the range, so clipping to a maximum energy such as maxE=-3dB
     makes sense:

     S = max(S, 10^(minE/10)); S = min(S, 10^(maxE/10));

     The frequency range of the FFT is from 0 to the Nyquist frequency
     of one half the sampling rate.  If the signal of interest is band
     limited, you do not need to display the entire frequency range.  In
     speech for example, most of the signal is below 4 kHz, so there is
     no reason to display up to the Nyquist frequency of 10 kHz for a 20
     kHz sampling rate.  In this case you will want to keep only the
     first 40% of the rows of the returned S and f.  More generally, to
     display the frequency range [minF, maxF], you could use the
     following row index:

     idx = (f >= minF & f <= maxF);

     Then there is the choice of colormap.  A brightness varying
     colormap such as copper or bone gives good shape to the ridges and
     valleys.  A hue varying colormap such as jet or hsv gives an
     indication of the steepness of the slopes.  The final spectrogram
     is displayed in log energy scale and by convention has low
     frequencies on the bottom of the image:

     imagesc(t, f, flipud(log(S(idx,:))));


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 40
Generate a spectrogram for the signal X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
square


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 379
 -- Function File: S = square (T, DUTY)
 -- Function File: S = square (T)
     Generate a square wave of period 2 pi with limits +1/-1.

     If DUTY is specified, the square wave is +1 for that portion of the
     time.

                         on time
        duty cycle = ------------------
                     on time + off time

     See also: cos, sawtooth, sin, tripuls.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 56
Generate a square wave of period 2 pi with limits +1/-1.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
ss2tf


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 351
 -- Function File: [NUM, DEN] = ss2tf (A, B, C, D)
     Conversion from transfer function to state-space.  The state space
     system:
                .
                x = Ax + Bu
                y = Cx + Du

     is converted to a transfer function:
                          num(s)
                    G(s)=-------
                          den(s)


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 49
Conversion from transfer function to state-space.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
ss2zp


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 165
 -- Function File: [P, Z, K] = ss2zp (A, B, C, D)
     Converts a state space representation to a set of poles and zeros;
     K is a gain associated with the zeros.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Converts a state space representation to a set of poles and zeros; K is
a gain a



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
tf2sos


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1288
 -- Function File: [SOS, G] = tf2sos (B, A)
 -- Function File: SOS = tf2sos (B, A)
     Convert direct-form filter coefficients to series second-order
     sections.

     INPUTS:

     B and A are vectors specifying the digital filter H(z) = B(z)/A(z).
     See 'filter' for documentation of the B and A filter coefficients.

     RETURNED:
        * SOS = matrix of series second-order sections, one per row:
               SOS = [B1.' A1.'; ...; BN.' AN.']
          where 'B1.' = [b0 b1 b2] and A1.' = [1 a1 a2]' for section 1,
          etc.  The b0 entry must be nonzero for each section (zeros at
          infinity not supported).
        * G is an overall gain factor that effectively scales any one of
          the Bi vectors.

     If called with only one output argument, the overall filter gain is
     applied to the first second-order section in the matrix SOS.

     EXAMPLE:
          B = [1 0 0 0 0 1];
          A = [1 0 0 0 0 .9];
          [sos, g] = tf2sos (B, A)

          sos =

             1.00000   0.61803   1.00000   1.00000   0.60515   0.95873
             1.00000  -1.61803   1.00000   1.00000  -1.58430   0.95873
             1.00000   1.00000  -0.00000   1.00000   0.97915  -0.00000

          g = 1

     See also: sos2tf, zp2sos, sos2pz, zp2tf, tf2zp.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 72
Convert direct-form filter coefficients to series second-order sections.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
tf2ss


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 516
 -- Function File: [A, B, C, D] = tf2ss (NUM, DEN)
     Conversion from transfer function to state-space.  The state space
     system:
                .
                x = Ax + Bu
                y = Cx + Du
     is obtained from a transfer function:
                          num(s)
                    G(s)=-------
                          den(s)

     The state space system matrices obtained from this function will be
     in observable companion form as Wolovich's Observable Structure
     Theorem is used.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 49
Conversion from transfer function to state-space.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
tf2zp


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 234
 -- Function File: [Z, P, K] = tf2zp (NUM, DEN)
     Convert transfer functions to poles-and-zero representations.

     Returns the zeros and poles of the system defined by NUM/DEN.  K is
     a gain associated with the system zeros.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 61
Convert transfer functions to poles-and-zero representations.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3
tfe


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 381
 Usage:
   [Pxx,freq] = tfe(x,y,Nfft,Fs,window,overlap,range,plot_type,detrend)

     Estimate transfer function of system with input "x" and output "y".
     Use the Welch (1967) periodogram/FFT method.
     Compatible with Matlab R11 tfe and earlier.
     See "help pwelch" for description of arguments, hints and references
     --- especially hint (7) for Matlab R11 defaults.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 79
 Usage:
   [Pxx,freq] = tfe(x,y,Nfft,Fs,window,overlap,range,plot_type,detrend)



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 10
tfestimate


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 524
 -- Function File: tfestimate (X, Y)
 -- Function File: tfestimate (X, Y, WINDOW)
 -- Function File: tfestimate (X, Y, WINDOW, OVERLAP)
 -- Function File: tfestimate (X, Y, WINDOW, OVERLAP, NFFT)
 -- Function File: tfestimate (X, Y, WINDOW, OVERLAP, NFFT, FS)
 -- Function File: tfestimate (X, Y, WINDOW, OVERLAP, NFFT, FS, RANGE)
 -- Function File: [PXX, FREQ] = tfestimate (...)

     Estimate transfer function of system with input X and output Y.
     Use the Welch (1967) periodogram/FFT method.

     See also: pwelch.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 63
Estimate transfer function of system with input X and output Y.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
triang


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 333
 -- Function File: triang (M)

     Return the filter coefficients of a triangular window of length M.
     Unlike the Bartlett window, 'triang' does not go to zero at the
     edges of the window.  For odd M, 'triang (M)' is equal to 'bartlett
     (M + 2)' except for the zeros at the edges of the window.

     See also: bartlett.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 66
Return the filter coefficients of a triangular window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
tripuls


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 775
 -- Function File: Y = tripuls (T)
 -- Function File: Y = tripuls (T, W)
 -- Function File: Y = tripuls (T, W, SKEW)

     Generate a triangular pulse over the interval [-w/2,w/2), sampled
     at times t.  This is useful with the function pulstran for
     generating a series pulses.

     skew is a value between -1 and 1, indicating the relative placement
     of the peak within the width.  -1 indicates that the peak should be
     at -w/2, and 1 indicates that the peak should be at w/2.

     Example:
          fs = 11025;  # arbitrary sample rate
          f0 = 100;    # pulse train sample rate
          w = 0.3/f0;  # pulse width 3/10th the distance between pulses
          auplot(pulstran(0:1/fs:4/f0, 0:1/f0:4/f0, 'tripuls', w), fs);

     See also: pulstran.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 77
Generate a triangular pulse over the interval [-w/2,w/2), sampled at
times t.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
tukeywin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 706
 -- Function File: tukeywin (M)
 -- Function File: tukeywin (M, R)
     Return the filter coefficients of a Tukey window (also known as the
     cosine-tapered window) of length M.  R defines the ratio between
     the constant section and and the cosine section.  It has to be
     between 0 and 1.  The function returns a Hanning window for R equal
     to 0 and a full box for R equals to 1.  The default value of R is
     1/2.

     For a definition of the Tukey window, see e.g.  Fredric J. Harris,
     "On the Use of Windows for Harmonic Analysis with the Discrete
     Fourier Transform, Proceedings of the IEEE", Vol.  66, No.  1,
     January 1978, Page 67, Equation 38.

     See also: hanning.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Return the filter coefficients of a Tukey window (also known as the
cosine-taper



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
ultrwin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 2418
 -- Function File: [W, XMU] = ultrwin (M, MU, BETA)
 -- Function File: [W, XMU] = ultrwin (M, MU, ATT, "att")
 -- Function File: [W, XMU] = ultrwin (M, MU, LATT, "latt")
 -- Function File: W = ultrwin (M, MU, XMU, "xmu")
     Return the coefficients of an Ultraspherical window of length M.
     The parameter MU controls the window's Fourier transform's
     side-lobe to side-lobe ratio, and the third given parameter
     controls the transform's main-lobe width/side-lobe-ratio; normalize
     W such that the central coefficient(s) value is unitary.

     By default, the third parameter is BETA, which sets the main lobe
     width to BETA times that of a rectangular window.  Alternatively,
     giving ATT or LATT sets the ripple ratio at the first or last
     side-lobe respectively, or giving XMU sets the (un-normalized)
     window's Fourier transform according to its canonical definition:

                  (MU)
          W(k) = C   [ XMU cos(pi k/M) ],  k = 0, 1, ..., M-1,
                  M-1

     where C is the Ultraspherical (a.k.a.  Gegenbauer) polynomial,
     which can be defined using the recurrence relationship:

           (l)    1                  (l)                    (l)
          C (x) = - [ 2x(m + l - 1) C   (x) - (m + 2l - 2) C   (x) ]
           m      m                  m-1                    m-2
     
                                     (l)        (l)
          for m an integer > 1, and C (x) = 1, C (x) = 2lx.
                                     0          1

     For given BETA, ATT, or LATT, the corresponding (determined) value
     of XMU is also returned.

     The Dolph-Chebyshev and Saramaki windows are special cases of the
     Ultraspherical window, with MU set to 0 and 1 respectively.  Note
     that when not giving XMU, stability issues may occur with MU <=
     -1.5.  For further information about the window, see

        * Kabal, P., 2009: Time Windows for Linear Prediction of Speech.
          Technical Report, Dept.  Elec.  & Comp.  Eng., McGill
          University.
        * Bergen, S., Antoniou, A., 2004: Design of Ultraspherical
          Window Functions with Prescribed Spectral Characteristics.
          Proc.  JASP, 13/13, pp.  2053-2065.
        * Streit, R., 1984: A two-parameter family of weights for
          nonrecursive digital filters and antennas.  Trans.  ASSP, 32,
          pp.  108-118.

     See also: chebwin, kaiser.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 64
Return the coefficients of an Ultraspherical window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
upsample


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 370
 -- Function File: Y = upsample (X, N)
 -- Function File: Y = upsample (X, N, OFFSET)
     Upsample the signal, inserting N-1 zeros between every element.

     If X is a matrix, upsample every column.

     If OFFSET is specified, control the position of the inserted sample
     in the block of N zeros.

     See also: decimate, downsample, interp, resample, upfirdn.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 63
Upsample the signal, inserting N-1 zeros between every element.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 12
upsamplefill


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 773
 -- Function File: Y = upsamplefill (X, V)
 -- Function File: Y = upsamplefill (..., COPY)
     Upsamples a vector interleaving given values or copies of the
     vector elements.

     The values in the vector V are placed between the elements of X.

     If the optional argument COPY is TRUE then V should be a scalar and
     each value in X are repeat V times.

     Example:
          upsamplefill (eye (2), 2, true)
          =>  1   0
              1   0
              1   0
              0   1
              0   1
              0   1
          upsamplefill (eye (2), [-1 -1 -1])
          =>  1   0
             -1  -1
             -1  -1
             -1  -1
              0   1
             -1  -1
             -1  -1
             -1  -1

     See also: upsample.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 78
Upsamples a vector interleaving given values or copies of the vector
elements.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
wconv


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 342
 -- Function File: Y = wconv (TYPE, X, F)
 -- Function File: Y = wconv (TYPE, X, F, SHAPE)
     1-D or 2-D convolution.

     *Inputs*
     TYPE
          Type of convolution.
     X
          Signal vector or matrix.
     F
          Coefficients of FIR filter.
     SHAPE
          Shape.

     *Outputs*
     Y
          Convoluted signal.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 23
1-D or 2-D convolution.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 8
welchwin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 794
 -- Function File: welchwin (M)
 -- Function File: welchwin (M, "periodic")
 -- Function File: welchwin (M, "symmetric")
     Return the filter coefficients of a Welch window of length M.  The
     Welch window is given by W(n)=1-(n/N-1)^2, n=[0,1, ...  M-1].  The
     optional argument specifies a "symmetric" window (the default) or a
     "periodic" window.

     A symmetric window has zero at each end and maximum in the middle,
     and the length must be an integer greater than 2.  The variable N
     in the formula above is '(M-1)/2'.

     A periodic window wraps around the cyclic interval [0,1, ...  M-1],
     and is intended for use with the DFT. The length must be an integer
     greater than 1.  The variable N in the formula above is 'M/2'.

     See also: blackman, kaiser.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 61
Return the filter coefficients of a Welch window of length M.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
window


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 256
 -- Function File: W = window (F, M)
 -- Function File: W = window (F, M, OPTS)
     Create an M-point window from the function F.  The function F can
     be for example '@blackman'.  Any additional arguments OPT are
     passed to the windowing function.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 45
Create an M-point window from the function F.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
wkeep


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 162
 -- Function File: Y = wkeep (X, L)
 -- Function File: Y = wkeep (X, L, OPT)
     Extract the elements of X of size L from the center, the right or
     the left.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 75
Extract the elements of X of size L from the center, the right or the
left.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
wrev


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 118
 -- Function File: Y = wrev (X)
     Reverse the order of the element of the vector X.

     See also: flipud, fliplr.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 49
Reverse the order of the element of the vector X.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5
xcorr


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 3021
 -- Function File: [R, LAG] = xcorr ( X )
 -- Function File: ... = xcorr ( X, Y )
 -- Function File: ... = xcorr ( ..., MAXLAG)
 -- Function File: ... = xcorr ( ..., SCALE)
     Estimates the cross-correlation.

     Estimate the cross correlation R_xy(k) of vector arguments X and Y
     or, if Y is omitted, estimate autocorrelation R_xx(k) of vector X,
     for a range of lags k specified by argument "maxlag".  If X is a
     matrix, each column of X is correlated with itself and every other
     column.

     The cross-correlation estimate between vectors "x" and "y" (of
     length N) for lag "k" is given by

                     N
          R_xy(k) = sum x_{i+k} conj(y_i),
                    i=1

     where data not provided (for example x(-1), y(N+1)) is zero.  Note
     the definition of cross-correlation given above.  To compute a
     cross-correlation consistent with the field of statistics, see
     'xcov'.

     *ARGUMENTS*
     X
          [non-empty; real or complex; vector or matrix] data

     Y
          [real or complex vector] data

          If X is a matrix (not a vector), Y must be omitted.  Y may be
          omitted if X is a vector; in this case xcorr estimates the
          autocorrelation of X.

     MAXLAG
          [integer scalar] maximum correlation lag If omitted, the
          default value is N-1, where N is the greater of the lengths of
          X and Y or, if X is a matrix, the number of rows in X.

     SCALE
          [character string] specifies the type of scaling applied to
          the correlation vector (or matrix).  is one of:
          'none'
               return the unscaled correlation, R,
          'biased'
               return the biased average, R/N,
          'unbiased'
               return the unbiased average, R(k)/(N-|k|),
          'coeff'
               return the correlation coefficient, R/(rms(x).rms(y)),
               where "k" is the lag, and "N" is the length of X.  If
               omitted, the default value is "none".  If Y is supplied
               but does not have the same length as X, scale must be
               "none".

     *RETURNED VARIABLES*
     R
          array of correlation estimates
     LAG
          row vector of correlation lags [-maxlag:maxlag]

     The array of correlation estimates has one of the following forms:
     (1) Cross-correlation estimate if X and Y are vectors.

     (2) Autocorrelation estimate if is a vector and Y is omitted.

     (3) If X is a matrix, R is an matrix containing the
     cross-correlation estimate of each column with every other column.
     Lag varies with the first index so that R has 2*maxlag+1 rows and
     P^2 columns where P is the number of columns in X.

     If Rij(k) is the correlation between columns i and j of X

     'R(k+maxlag+1,P*(i-1)+j) == Rij(k)'

     for lag k in [-maxlag:maxlag], or

     'R(:,P*(i-1)+j) == xcorr(X(:,i),X(:,j))'.

     'reshape(R(k,:),P,P)' is the cross-correlation matrix for 'X(k,:)'.

     See also: xcov.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 32
Estimates the cross-correlation.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
xcorr2


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1009
 -- Function File: xcorr2 (A)
 -- Function File: xcorr2 (A, B)
 -- Function File: xcorr2 (..., SCALE)
     Compute the 2D cross-correlation of matrices A and B.

     If B is not specified, computes autocorrelation of A, i.e., same as
     'xcorr (A, A)'.

     The optional argument SCALE, defines the type of scaling applied to
     the cross-correlation matrix.  Possible values are:

     "none" (default)
          No scaling.

     "biased"
          Scales the raw cross-correlation by the maximum number of
          elements of A and B involved in the generation of any element
          of C.

     "unbiased"
          Scales the raw correlation by dividing each element in the
          cross-correlation matrix by the number of products A and B
          used to generate that element.

     "coeff"
          Scales the normalized cross-correlation on the range of [0 1]
          so that a value of 1 corresponds to a correlation coefficient
          of 1.

     See also: conv2, corr2, xcorr.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 53
Compute the 2D cross-correlation of matrices A and B.



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4
xcov


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 938
 -- Function File: [R, LAG] = xcov ( X )
 -- Function File: ... = xcov ( X, Y )
 -- Function File: ... = xcov ( ..., MAXLAG)
 -- Function File: ... = xcov ( ..., SCALE)
     Compute covariance at various lags
     [=correlation(x-mean(x),y-mean(y))].

     X
          input vector
     Y
          if specified, compute cross-covariance between X and Y,
          otherwise compute autocovariance of X.
     MAXLAG
          is specified, use lag range [-maxlag:maxlag], otherwise use
          range [-n+1:n-1].
     SCALE:
          'biased'
               for covariance=raw/N,
          'unbiased'
               for covariance=raw/(N-|lag|),
          'coeff'
               for covariance=raw/(covariance at lag 0),
          'none'
               for covariance=raw
          'none'
               is the default.

     Returns the covariance for each lag in the range, plus an optional
     vector of lags.

     See also: xcorr.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 71
Compute covariance at various lags [=correlation(x-mean(x),y-mean(y))].



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 12
zerocrossing


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 184
 -- Function File: X0 = zerocrossing (X, Y)
     Estimates the points at which a given waveform y=y(x) crosses the
     x-axis using linear interpolation.

     See also: fzero, roots.


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
Estimates the points at which a given waveform y=y(x) crosses the x-axis
using l



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 6
zp2sos


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 1513
 -- Function File: [SOS, G] = zp2sos (Z)
 -- Function File: [SOS, G] = zp2sos (Z, P)
 -- Function File: [SOS, G] = zp2sos (Z, P, K)
 -- Function File: SOS = zp2sos (...)
     Convert filter poles and zeros to second-order sections.

     INPUTS:
        * Z = column-vector containing the filter zeros
        * P = column-vector containing the filter poles
        * K = overall filter gain factor If not given the gain is
          assumed to be 1.

     RETURNED:
        * SOS = matrix of series second-order sections, one per row:
               SOS = [B1.' A1.'; ...; BN.' AN.']
          where 'B1.' = [b0 b1 b2] and A1.' = [1 a1 a2]' for section 1,
          etc.  The b0 entry must be nonzero for each section.  See
          'filter' for documentation of the second-order direct-form
          filter coefficients Bi and %Ai, i=1:N.

        * G is the overall gain factor that effectively scales any one
          of the Bi vectors.

     If called with only one output argument, the overall filter gain is
     applied to the first second-order section in the matrix SOS.

     EXAMPLE:
            [z, p, k] = tf2zp ([1 0 0 0 0 1], [1 0 0 0 0 .9]);
            [sos, g] = zp2sos (z, p, k)

          sos =
             1.0000    0.6180    1.0000    1.0000    0.6051    0.9587
             1.0000   -1.6180    1.0000    1.0000   -1.5843    0.9587
             1.0000    1.0000         0    1.0000    0.9791         0

          g =
              1

     See also: sos2pz, sos2tf, tf2sos, zp2tf, tf2zp.


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Convert filter poles and zeros to second-order sections.



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zp2ss


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 -- Function File: [A, B, C, D] = zp2ss (Z, P, K)
     Conversion from zero / pole to state space.

     *Inputs*
     Z
     P
          Vectors of (possibly) complex poles and zeros of a transfer
          function.  Complex values must come in conjugate pairs (i.e.,
          x+jy in Z means that x-jy is also in Z).
     K
          Real scalar (leading coefficient).

     *Outputs*
     A
     B
     C
     D
          The state space system, in the form:
                    .
                    x = Ax + Bu
                    y = Cx + Du


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Conversion from zero / pole to state space.



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zp2tf


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 -- Function File: [NUM, DEN] = zp2tf (Z, P, K)
     Converts zeros / poles to a transfer function.

     *Inputs*
     Z
     P
          Vectors of (possibly complex) poles and zeros of a transfer
          function.  Complex values must appear in conjugate pairs.
     K
          Real scalar (leading coefficient).


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Converts zeros / poles to a transfer function.



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zplane


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 -- Function File: zplane (Z, P)
 -- Function File: zplane (B, A)

     Plot the poles and zeros.  If the arguments are row vectors then
     they represent filter coefficients (numerator polynomial b and
     denominator polynomial a), but if they are column vectors or
     matrices then they represent poles and zeros.

     This is a horrid interface, but I didn't choose it; better would be
     to accept b,a or z,p,g like other functions.  The saving grace is
     that poly(x) always returns a row vector and roots(x) always
     returns a column vector, so it is usually right.  You must only be
     careful when you are creating filters by hand.

     Note that due to the nature of the roots() function, poles and
     zeros may be displayed as occurring around a circle rather than at
     a single point.

     The transfer function is

                 B(z)   b0 + b1 z^(-1) + b2 z^(-2) + ... + bM z^(-M)
          H(z) = ---- = --------------------------------------------
                 A(z)   a0 + a1 z^(-1) + a2 z^(-2) + ... + aN z^(-N)

                        b0          (z - z1) (z - z2) ... (z - zM)
                      = -- z^(-M+N) ------------------------------
                        a0          (z - p1) (z - p2) ... (z - pN)

     The denominator a defaults to 1, and the poles p defaults to [].


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Plot the poles and zeros.