package scipy

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val get_py : string -> Py.Object.t

Get an attribute of this module as a Py.Object.t. This is useful to pass a Python function to another function.

module CKDTree : sig ... end
module Sp_fft : sig ... end
val axis_reverse : ?axis:Py.Object.t -> a:Py.Object.t -> unit -> Py.Object.t

Reverse the 1-d slices of `a` along axis `axis`.

Returns axis_slice(a, step=-1, axis=axis).

val axis_slice : ?start:Py.Object.t -> ?stop:Py.Object.t -> ?step:Py.Object.t -> ?axis:int -> a:[> `Ndarray ] Np.Obj.t -> unit -> Py.Object.t

Take a slice along axis 'axis' from 'a'.

Parameters ---------- a : numpy.ndarray The array to be sliced. start, stop, step : int or None The slice parameters. axis : int, optional The axis of `a` to be sliced.

Examples -------- >>> a = array([1, 2, 3], [4, 5, 6], [7, 8, 9]) >>> axis_slice(a, start=0, stop=1, axis=1) array([1], [4], [7]) >>> axis_slice(a, start=1, axis=0) array([4, 5, 6], [7, 8, 9])

Notes ----- The keyword arguments start, stop and step are used by calling slice(start, stop, step). This implies axis_slice() does not handle its arguments the exactly the same as indexing. To select a single index k, for example, use axis_slice(a, start=k, stop=k+1) In this case, the length of the axis 'axis' in the result will be 1; the trivial dimension is not removed. (Use numpy.squeeze() to remove trivial axes.)

val callable : Py.Object.t -> Py.Object.t

None

val cheby1 : ?btype:[ `Lowpass | `Highpass | `Bandpass | `Bandstop ] -> ?analog:bool -> ?output:[ `Ba | `Zpk | `Sos ] -> ?fs:float -> n:int -> rp:float -> wn:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Chebyshev type I digital and analog filter design.

Design an Nth-order digital or analog Chebyshev type I filter and return the filter coefficients.

Parameters ---------- N : int The order of the filter. rp : float The maximum ripple allowed below unity gain in the passband. Specified in decibels, as a positive number. Wn : array_like A scalar or length-2 sequence giving the critical frequencies. For Type I filters, this is the point in the transition band at which the gain first drops below -`rp`.

For digital filters, `Wn` are in the same units as `fs`. By default, `fs` is 2 half-cycles/sample, so these are normalized from 0 to 1, where 1 is the Nyquist frequency. (`Wn` is thus in half-cycles / sample.)

For analog filters, `Wn` is an angular frequency (e.g. rad/s). btype : 'lowpass', 'highpass', 'bandpass', 'bandstop', optional The type of filter. Default is 'lowpass'. analog : bool, optional When True, return an analog filter, otherwise a digital filter is returned. output : 'ba', 'zpk', 'sos', optional Type of output: numerator/denominator ('ba'), pole-zero ('zpk'), or second-order sections ('sos'). Default is 'ba' for backwards compatibility, but 'sos' should be used for general-purpose filtering. fs : float, optional The sampling frequency of the digital system.

.. versionadded:: 1.2.0

Returns ------- b, a : ndarray, ndarray Numerator (`b`) and denominator (`a`) polynomials of the IIR filter. Only returned if ``output='ba'``. z, p, k : ndarray, ndarray, float Zeros, poles, and system gain of the IIR filter transfer function. Only returned if ``output='zpk'``. sos : ndarray Second-order sections representation of the IIR filter. Only returned if ``output=='sos'``.

See Also -------- cheb1ord, cheb1ap

Notes ----- The Chebyshev type I filter maximizes the rate of cutoff between the frequency response's passband and stopband, at the expense of ripple in the passband and increased ringing in the step response.

Type I filters roll off faster than Type II (`cheby2`), but Type II filters do not have any ripple in the passband.

The equiripple passband has N maxima or minima (for example, a 5th-order filter has 3 maxima and 2 minima). Consequently, the DC gain is unity for odd-order filters, or -rp dB for even-order filters.

The ``'sos'`` output parameter was added in 0.16.0.

Examples -------- Design an analog filter and plot its frequency response, showing the critical points:

>>> from scipy import signal >>> import matplotlib.pyplot as plt

>>> b, a = signal.cheby1(4, 5, 100, 'low', analog=True) >>> w, h = signal.freqs(b, a) >>> plt.semilogx(w, 20 * np.log10(abs(h))) >>> plt.title('Chebyshev Type I frequency response (rp=5)') >>> plt.xlabel('Frequency radians / second') >>> plt.ylabel('Amplitude dB') >>> plt.margins(0, 0.1) >>> plt.grid(which='both', axis='both') >>> plt.axvline(100, color='green') # cutoff frequency >>> plt.axhline(-5, color='green') # rp >>> plt.show()

Generate a signal made up of 10 Hz and 20 Hz, sampled at 1 kHz

>>> t = np.linspace(0, 1, 1000, False) # 1 second >>> sig = np.sin(2*np.pi*10*t) + np.sin(2*np.pi*20*t) >>> fig, (ax1, ax2) = plt.subplots(2, 1, sharex=True) >>> ax1.plot(t, sig) >>> ax1.set_title('10 Hz and 20 Hz sinusoids') >>> ax1.axis(0, 1, -2, 2)

Design a digital high-pass filter at 15 Hz to remove the 10 Hz tone, and apply it to the signal. (It's recommended to use second-order sections format when filtering, to avoid numerical error with transfer function (``ba``) format):

>>> sos = signal.cheby1(10, 1, 15, 'hp', fs=1000, output='sos') >>> filtered = signal.sosfilt(sos, sig) >>> ax2.plot(t, filtered) >>> ax2.set_title('After 15 Hz high-pass filter') >>> ax2.axis(0, 1, -2, 2) >>> ax2.set_xlabel('Time seconds') >>> plt.tight_layout() >>> plt.show()

val choose_conv_method : ?mode:[ `Full | `Valid | `Same ] -> ?measure:bool -> in1:[> `Ndarray ] Np.Obj.t -> in2:[> `Ndarray ] Np.Obj.t -> unit -> string * Py.Object.t

Find the fastest convolution/correlation method.

This primarily exists to be called during the ``method='auto'`` option in `convolve` and `correlate`. It can also be used to determine the value of ``method`` for many different convolutions of the same dtype/shape. In addition, it supports timing the convolution to adapt the value of ``method`` to a particular set of inputs and/or hardware.

Parameters ---------- in1 : array_like The first argument passed into the convolution function. in2 : array_like The second argument passed into the convolution function. mode : str 'full', 'valid', 'same', optional A string indicating the size of the output:

``full`` The output is the full discrete linear convolution of the inputs. (Default) ``valid`` The output consists only of those elements that do not rely on the zero-padding. ``same`` The output is the same size as `in1`, centered with respect to the 'full' output. measure : bool, optional If True, run and time the convolution of `in1` and `in2` with both methods and return the fastest. If False (default), predict the fastest method using precomputed values.

Returns ------- method : str A string indicating which convolution method is fastest, either 'direct' or 'fft' times : dict, optional A dictionary containing the times (in seconds) needed for each method. This value is only returned if ``measure=True``.

See Also -------- convolve correlate

Notes ----- Generally, this method is 99% accurate for 2D signals and 85% accurate for 1D signals for randomly chosen input sizes. For precision, use ``measure=True`` to find the fastest method by timing the convolution. This can be used to avoid the minimal overhead of finding the fastest ``method`` later, or to adapt the value of ``method`` to a particular set of inputs.

Experiments were run on an Amazon EC2 r5a.2xlarge machine to test this function. These experiments measured the ratio between the time required when using ``method='auto'`` and the time required for the fastest method (i.e., ``ratio = time_auto / min(time_fft, time_direct)``). In these experiments, we found:

* There is a 95% chance of this ratio being less than 1.5 for 1D signals and a 99% chance of being less than 2.5 for 2D signals. * The ratio was always less than 2.5/5 for 1D/2D signals respectively. * This function is most inaccurate for 1D convolutions that take between 1 and 10 milliseconds with ``method='direct'``. A good proxy for this (at least in our experiments) is ``1e6 <= in1.size * in2.size <= 1e7``.

The 2D results almost certainly generalize to 3D/4D/etc because the implementation is the same (the 1D implementation is different).

All the numbers above are specific to the EC2 machine. However, we did find that this function generalizes fairly decently across hardware. The speed tests were of similar quality (and even slightly better) than the same tests performed on the machine to tune this function's numbers (a mid-2014 15-inch MacBook Pro with 16GB RAM and a 2.5GHz Intel i7 processor).

There are cases when `fftconvolve` supports the inputs but this function returns `direct` (e.g., to protect against floating point integer precision).

.. versionadded:: 0.19

Examples -------- Estimate the fastest method for a given input:

>>> from scipy import signal >>> img = np.random.rand(32, 32) >>> filter = np.random.rand(8, 8) >>> method = signal.choose_conv_method(img, filter, mode='same') >>> method 'fft'

This can then be applied to other arrays of the same dtype and shape:

>>> img2 = np.random.rand(32, 32) >>> filter2 = np.random.rand(8, 8) >>> corr2 = signal.correlate(img2, filter2, mode='same', method=method) >>> conv2 = signal.convolve(img2, filter2, mode='same', method=method)

The output of this function (``method``) works with `correlate` and `convolve`.

val cmplx_sort : [> `Ndarray ] Np.Obj.t -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Sort roots based on magnitude.

Parameters ---------- p : array_like The roots to sort, as a 1-D array.

Returns ------- p_sorted : ndarray Sorted roots. indx : ndarray Array of indices needed to sort the input `p`.

Examples -------- >>> from scipy import signal >>> vals = 1, 4, 1+1.j, 3 >>> p_sorted, indx = signal.cmplx_sort(vals) >>> p_sorted array(1.+0.j, 1.+1.j, 3.+0.j, 4.+0.j) >>> indx array(0, 2, 3, 1)

val const_ext : ?axis:int -> x:[> `Ndarray ] Np.Obj.t -> n:int -> unit -> Py.Object.t

Constant extension at the boundaries of an array

Generate a new ndarray that is a constant extension of `x` along an axis.

The extension repeats the values at the first and last element of the axis.

Parameters ---------- x : ndarray The array to be extended. n : int The number of elements by which to extend `x` at each end of the axis. axis : int, optional The axis along which to extend `x`. Default is -1.

Examples -------- >>> from scipy.signal._arraytools import const_ext >>> a = np.array([1, 2, 3, 4, 5], [0, 1, 4, 9, 16]) >>> const_ext(a, 2) array([ 1, 1, 1, 2, 3, 4, 5, 5, 5], [ 0, 0, 0, 1, 4, 9, 16, 16, 16])

Constant extension continues with the same values as the endpoints of the array:

>>> t = np.linspace(0, 1.5, 100) >>> a = 0.9 * np.sin(2 * np.pi * t**2) >>> b = const_ext(a, 40) >>> import matplotlib.pyplot as plt >>> plt.plot(arange(-40, 140), b, 'b', lw=1, label='constant extension') >>> plt.plot(arange(100), a, 'r', lw=2, label='original') >>> plt.legend(loc='best') >>> plt.show()

val convolve : ?mode:[ `Full | `Valid | `Same ] -> ?method_:[ `Auto | `Direct | `Fft ] -> in1:[> `Ndarray ] Np.Obj.t -> in2:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Convolve two N-dimensional arrays.

Convolve `in1` and `in2`, with the output size determined by the `mode` argument.

Parameters ---------- in1 : array_like First input. in2 : array_like Second input. Should have the same number of dimensions as `in1`. mode : str 'full', 'valid', 'same', optional A string indicating the size of the output:

``full`` The output is the full discrete linear convolution of the inputs. (Default) ``valid`` The output consists only of those elements that do not rely on the zero-padding. In 'valid' mode, either `in1` or `in2` must be at least as large as the other in every dimension. ``same`` The output is the same size as `in1`, centered with respect to the 'full' output. method : str 'auto', 'direct', 'fft', optional A string indicating which method to use to calculate the convolution.

``direct`` The convolution is determined directly from sums, the definition of convolution. ``fft`` The Fourier Transform is used to perform the convolution by calling `fftconvolve`. ``auto`` Automatically chooses direct or Fourier method based on an estimate of which is faster (default). See Notes for more detail.

.. versionadded:: 0.19.0

Returns ------- convolve : array An N-dimensional array containing a subset of the discrete linear convolution of `in1` with `in2`.

See Also -------- numpy.polymul : performs polynomial multiplication (same operation, but also accepts poly1d objects) choose_conv_method : chooses the fastest appropriate convolution method fftconvolve : Always uses the FFT method. oaconvolve : Uses the overlap-add method to do convolution, which is generally faster when the input arrays are large and significantly different in size.

Notes ----- By default, `convolve` and `correlate` use ``method='auto'``, which calls `choose_conv_method` to choose the fastest method using pre-computed values (`choose_conv_method` can also measure real-world timing with a keyword argument). Because `fftconvolve` relies on floating point numbers, there are certain constraints that may force `method=direct` (more detail in `choose_conv_method` docstring).

Examples -------- Smooth a square pulse using a Hann window:

>>> from scipy import signal >>> sig = np.repeat(0., 1., 0., 100) >>> win = signal.hann(50) >>> filtered = signal.convolve(sig, win, mode='same') / sum(win)

>>> import matplotlib.pyplot as plt >>> fig, (ax_orig, ax_win, ax_filt) = plt.subplots(3, 1, sharex=True) >>> ax_orig.plot(sig) >>> ax_orig.set_title('Original pulse') >>> ax_orig.margins(0, 0.1) >>> ax_win.plot(win) >>> ax_win.set_title('Filter impulse response') >>> ax_win.margins(0, 0.1) >>> ax_filt.plot(filtered) >>> ax_filt.set_title('Filtered signal') >>> ax_filt.margins(0, 0.1) >>> fig.tight_layout() >>> fig.show()

val convolve2d : ?mode:[ `Full | `Valid | `Same ] -> ?boundary:[ `Fill | `Wrap | `Symm ] -> ?fillvalue:[ `F of float | `I of int | `Bool of bool | `S of string ] -> in1:[> `Ndarray ] Np.Obj.t -> in2:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Convolve two 2-dimensional arrays.

Convolve `in1` and `in2` with output size determined by `mode`, and boundary conditions determined by `boundary` and `fillvalue`.

Parameters ---------- in1 : array_like First input. in2 : array_like Second input. Should have the same number of dimensions as `in1`. mode : str 'full', 'valid', 'same', optional A string indicating the size of the output:

``full`` The output is the full discrete linear convolution of the inputs. (Default) ``valid`` The output consists only of those elements that do not rely on the zero-padding. In 'valid' mode, either `in1` or `in2` must be at least as large as the other in every dimension. ``same`` The output is the same size as `in1`, centered with respect to the 'full' output. boundary : str 'fill', 'wrap', 'symm', optional A flag indicating how to handle boundaries:

``fill`` pad input arrays with fillvalue. (default) ``wrap`` circular boundary conditions. ``symm`` symmetrical boundary conditions.

fillvalue : scalar, optional Value to fill pad input arrays with. Default is 0.

Returns ------- out : ndarray A 2-dimensional array containing a subset of the discrete linear convolution of `in1` with `in2`.

Examples -------- Compute the gradient of an image by 2D convolution with a complex Scharr operator. (Horizontal operator is real, vertical is imaginary.) Use symmetric boundary condition to avoid creating edges at the image boundaries.

>>> from scipy import signal >>> from scipy import misc >>> ascent = misc.ascent() >>> scharr = np.array([ -3-3j, 0-10j, +3 -3j], ... [-10+0j, 0+ 0j, +10 +0j], ... [ -3+3j, 0+10j, +3 +3j]) # Gx + j*Gy >>> grad = signal.convolve2d(ascent, scharr, boundary='symm', mode='same')

>>> import matplotlib.pyplot as plt >>> fig, (ax_orig, ax_mag, ax_ang) = plt.subplots(3, 1, figsize=(6, 15)) >>> ax_orig.imshow(ascent, cmap='gray') >>> ax_orig.set_title('Original') >>> ax_orig.set_axis_off() >>> ax_mag.imshow(np.absolute(grad), cmap='gray') >>> ax_mag.set_title('Gradient magnitude') >>> ax_mag.set_axis_off() >>> ax_ang.imshow(np.angle(grad), cmap='hsv') # hsv is cyclic, like angles >>> ax_ang.set_title('Gradient orientation') >>> ax_ang.set_axis_off() >>> fig.show()

val correlate : ?mode:[ `Full | `Valid | `Same ] -> ?method_:[ `Auto | `Direct | `Fft ] -> in1:[> `Ndarray ] Np.Obj.t -> in2:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Cross-correlate two N-dimensional arrays.

Cross-correlate `in1` and `in2`, with the output size determined by the `mode` argument.

Parameters ---------- in1 : array_like First input. in2 : array_like Second input. Should have the same number of dimensions as `in1`. mode : str 'full', 'valid', 'same', optional A string indicating the size of the output:

``full`` The output is the full discrete linear cross-correlation of the inputs. (Default) ``valid`` The output consists only of those elements that do not rely on the zero-padding. In 'valid' mode, either `in1` or `in2` must be at least as large as the other in every dimension. ``same`` The output is the same size as `in1`, centered with respect to the 'full' output. method : str 'auto', 'direct', 'fft', optional A string indicating which method to use to calculate the correlation.

``direct`` The correlation is determined directly from sums, the definition of correlation. ``fft`` The Fast Fourier Transform is used to perform the correlation more quickly (only available for numerical arrays.) ``auto`` Automatically chooses direct or Fourier method based on an estimate of which is faster (default). See `convolve` Notes for more detail.

.. versionadded:: 0.19.0

Returns ------- correlate : array An N-dimensional array containing a subset of the discrete linear cross-correlation of `in1` with `in2`.

See Also -------- choose_conv_method : contains more documentation on `method`.

Notes ----- The correlation z of two d-dimensional arrays x and y is defined as::

z...,k,... = sum..., i_l, ... x..., i_l,... * conj(y..., i_l - k,...)

This way, if x and y are 1-D arrays and ``z = correlate(x, y, 'full')`` then

.. math::

zk = (x * y)(k - N + 1) = \sum_l=0^ ||x||-1x_l y_l-k+N-1^*

for :math:`k = 0, 1, ..., ||x|| + ||y|| - 2`

where :math:`||x||` is the length of ``x``, :math:`N = \max(||x||,||y||)`, and :math:`y_m` is 0 when m is outside the range of y.

``method='fft'`` only works for numerical arrays as it relies on `fftconvolve`. In certain cases (i.e., arrays of objects or when rounding integers can lose precision), ``method='direct'`` is always used.

Examples -------- Implement a matched filter using cross-correlation, to recover a signal that has passed through a noisy channel.

>>> from scipy import signal >>> sig = np.repeat(0., 1., 1., 0., 1., 0., 0., 1., 128) >>> sig_noise = sig + np.random.randn(len(sig)) >>> corr = signal.correlate(sig_noise, np.ones(128), mode='same') / 128

>>> import matplotlib.pyplot as plt >>> clock = np.arange(64, len(sig), 128) >>> fig, (ax_orig, ax_noise, ax_corr) = plt.subplots(3, 1, sharex=True) >>> ax_orig.plot(sig) >>> ax_orig.plot(clock, sigclock, 'ro') >>> ax_orig.set_title('Original signal') >>> ax_noise.plot(sig_noise) >>> ax_noise.set_title('Signal with noise') >>> ax_corr.plot(corr) >>> ax_corr.plot(clock, corrclock, 'ro') >>> ax_corr.axhline(0.5, ls=':') >>> ax_corr.set_title('Cross-correlated with rectangular pulse') >>> ax_orig.margins(0, 0.1) >>> fig.tight_layout() >>> fig.show()

val correlate2d : ?mode:[ `Full | `Valid | `Same ] -> ?boundary:[ `Fill | `Wrap | `Symm ] -> ?fillvalue:[ `F of float | `I of int | `Bool of bool | `S of string ] -> in1:[> `Ndarray ] Np.Obj.t -> in2:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Cross-correlate two 2-dimensional arrays.

Cross correlate `in1` and `in2` with output size determined by `mode`, and boundary conditions determined by `boundary` and `fillvalue`.

Parameters ---------- in1 : array_like First input. in2 : array_like Second input. Should have the same number of dimensions as `in1`. mode : str 'full', 'valid', 'same', optional A string indicating the size of the output:

``full`` The output is the full discrete linear cross-correlation of the inputs. (Default) ``valid`` The output consists only of those elements that do not rely on the zero-padding. In 'valid' mode, either `in1` or `in2` must be at least as large as the other in every dimension. ``same`` The output is the same size as `in1`, centered with respect to the 'full' output. boundary : str 'fill', 'wrap', 'symm', optional A flag indicating how to handle boundaries:

``fill`` pad input arrays with fillvalue. (default) ``wrap`` circular boundary conditions. ``symm`` symmetrical boundary conditions.

fillvalue : scalar, optional Value to fill pad input arrays with. Default is 0.

Returns ------- correlate2d : ndarray A 2-dimensional array containing a subset of the discrete linear cross-correlation of `in1` with `in2`.

Examples -------- Use 2D cross-correlation to find the location of a template in a noisy image:

>>> from scipy import signal >>> from scipy import misc >>> face = misc.face(gray=True) - misc.face(gray=True).mean() >>> template = np.copy(face300:365, 670:750) # right eye >>> template -= template.mean() >>> face = face + np.random.randn( *face.shape) * 50 # add noise >>> corr = signal.correlate2d(face, template, boundary='symm', mode='same') >>> y, x = np.unravel_index(np.argmax(corr), corr.shape) # find the match

>>> import matplotlib.pyplot as plt >>> fig, (ax_orig, ax_template, ax_corr) = plt.subplots(3, 1, ... figsize=(6, 15)) >>> ax_orig.imshow(face, cmap='gray') >>> ax_orig.set_title('Original') >>> ax_orig.set_axis_off() >>> ax_template.imshow(template, cmap='gray') >>> ax_template.set_title('Template') >>> ax_template.set_axis_off() >>> ax_corr.imshow(corr, cmap='gray') >>> ax_corr.set_title('Cross-correlation') >>> ax_corr.set_axis_off() >>> ax_orig.plot(x, y, 'ro') >>> fig.show()

val decimate : ?n:int -> ?ftype:[ `Fir | `Iir | `T_dlti_instance of Py.Object.t ] -> ?axis:int -> ?zero_phase:bool -> x:[> `Ndarray ] Np.Obj.t -> q:int -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Downsample the signal after applying an anti-aliasing filter.

By default, an order 8 Chebyshev type I filter is used. A 30 point FIR filter with Hamming window is used if `ftype` is 'fir'.

Parameters ---------- x : array_like The signal to be downsampled, as an N-dimensional array. q : int The downsampling factor. When using IIR downsampling, it is recommended to call `decimate` multiple times for downsampling factors higher than 13. n : int, optional The order of the filter (1 less than the length for 'fir'). Defaults to 8 for 'iir' and 20 times the downsampling factor for 'fir'. ftype : str 'iir', 'fir' or ``dlti`` instance, optional If 'iir' or 'fir', specifies the type of lowpass filter. If an instance of an `dlti` object, uses that object to filter before downsampling. axis : int, optional The axis along which to decimate. zero_phase : bool, optional Prevent phase shift by filtering with `filtfilt` instead of `lfilter` when using an IIR filter, and shifting the outputs back by the filter's group delay when using an FIR filter. The default value of ``True`` is recommended, since a phase shift is generally not desired.

.. versionadded:: 0.18.0

Returns ------- y : ndarray The down-sampled signal.

See Also -------- resample : Resample up or down using the FFT method. resample_poly : Resample using polyphase filtering and an FIR filter.

Notes ----- The ``zero_phase`` keyword was added in 0.18.0. The possibility to use instances of ``dlti`` as ``ftype`` was added in 0.18.0.

val deconvolve : signal:[> `Ndarray ] Np.Obj.t -> divisor:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Deconvolves ``divisor`` out of ``signal`` using inverse filtering.

Returns the quotient and remainder such that ``signal = convolve(divisor, quotient) + remainder``

Parameters ---------- signal : array_like Signal data, typically a recorded signal divisor : array_like Divisor data, typically an impulse response or filter that was applied to the original signal

Returns ------- quotient : ndarray Quotient, typically the recovered original signal remainder : ndarray Remainder

Examples -------- Deconvolve a signal that's been filtered:

>>> from scipy import signal >>> original = 0, 1, 0, 0, 1, 1, 0, 0 >>> impulse_response = 2, 1 >>> recorded = signal.convolve(impulse_response, original) >>> recorded array(0, 2, 1, 0, 2, 3, 1, 0, 0) >>> recovered, remainder = signal.deconvolve(recorded, impulse_response) >>> recovered array( 0., 1., 0., 0., 1., 1., 0., 0.)

See Also -------- numpy.polydiv : performs polynomial division (same operation, but also accepts poly1d objects)

val detrend : ?axis:int -> ?type_:[ `Linear | `Constant ] -> ?bp:Py.Object.t -> ?overwrite_data:bool -> data:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Remove linear trend along axis from data.

Parameters ---------- data : array_like The input data. axis : int, optional The axis along which to detrend the data. By default this is the last axis (-1). type : 'linear', 'constant', optional The type of detrending. If ``type == 'linear'`` (default), the result of a linear least-squares fit to `data` is subtracted from `data`. If ``type == 'constant'``, only the mean of `data` is subtracted. bp : array_like of ints, optional A sequence of break points. If given, an individual linear fit is performed for each part of `data` between two break points. Break points are specified as indices into `data`. overwrite_data : bool, optional If True, perform in place detrending and avoid a copy. Default is False

Returns ------- ret : ndarray The detrended input data.

Examples -------- >>> from scipy import signal >>> randgen = np.random.RandomState(9) >>> npoints = 1000 >>> noise = randgen.randn(npoints) >>> x = 3 + 2*np.linspace(0, 1, npoints) + noise >>> (signal.detrend(x) - noise).max() < 0.01 True

val even_ext : ?axis:int -> x:[> `Ndarray ] Np.Obj.t -> n:int -> unit -> Py.Object.t

Even extension at the boundaries of an array

Generate a new ndarray by making an even extension of `x` along an axis.

Parameters ---------- x : ndarray The array to be extended. n : int The number of elements by which to extend `x` at each end of the axis. axis : int, optional The axis along which to extend `x`. Default is -1.

Examples -------- >>> from scipy.signal._arraytools import even_ext >>> a = np.array([1, 2, 3, 4, 5], [0, 1, 4, 9, 16]) >>> even_ext(a, 2) array([ 3, 2, 1, 2, 3, 4, 5, 4, 3], [ 4, 1, 0, 1, 4, 9, 16, 9, 4])

Even extension is a 'mirror image' at the boundaries of the original array:

>>> t = np.linspace(0, 1.5, 100) >>> a = 0.9 * np.sin(2 * np.pi * t**2) >>> b = even_ext(a, 40) >>> import matplotlib.pyplot as plt >>> plt.plot(arange(-40, 140), b, 'b', lw=1, label='even extension') >>> plt.plot(arange(100), a, 'r', lw=2, label='original') >>> plt.legend(loc='best') >>> plt.show()

val factorial : ?exact:bool -> n:[ `I of int | `Array_like_of_ints of Py.Object.t ] -> unit -> Py.Object.t

The factorial of a number or array of numbers.

The factorial of non-negative integer `n` is the product of all positive integers less than or equal to `n`::

n! = n * (n - 1) * (n - 2) * ... * 1

Parameters ---------- n : int or array_like of ints Input values. If ``n < 0``, the return value is 0. exact : bool, optional If True, calculate the answer exactly using long integer arithmetic. If False, result is approximated in floating point rapidly using the `gamma` function. Default is False.

Returns ------- nf : float or int or ndarray Factorial of `n`, as integer or float depending on `exact`.

Notes ----- For arrays with ``exact=True``, the factorial is computed only once, for the largest input, with each other result computed in the process. The output dtype is increased to ``int64`` or ``object`` if necessary.

With ``exact=False`` the factorial is approximated using the gamma function:

.. math:: n! = \Gamma(n+1)

Examples -------- >>> from scipy.special import factorial >>> arr = np.array(3, 4, 5) >>> factorial(arr, exact=False) array( 6., 24., 120.) >>> factorial(arr, exact=True) array( 6, 24, 120) >>> factorial(5, exact=True) 120L

val fftconvolve : ?mode:[ `Full | `Valid | `Same ] -> ?axes:[ `I of int | `Array_like_of_ints of Py.Object.t ] -> in1:[> `Ndarray ] Np.Obj.t -> in2:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Convolve two N-dimensional arrays using FFT.

Convolve `in1` and `in2` using the fast Fourier transform method, with the output size determined by the `mode` argument.

This is generally much faster than `convolve` for large arrays (n > ~500), but can be slower when only a few output values are needed, and can only output float arrays (int or object array inputs will be cast to float).

As of v0.19, `convolve` automatically chooses this method or the direct method based on an estimation of which is faster.

Parameters ---------- in1 : array_like First input. in2 : array_like Second input. Should have the same number of dimensions as `in1`. mode : str 'full', 'valid', 'same', optional A string indicating the size of the output:

``full`` The output is the full discrete linear convolution of the inputs. (Default) ``valid`` The output consists only of those elements that do not rely on the zero-padding. In 'valid' mode, either `in1` or `in2` must be at least as large as the other in every dimension. ``same`` The output is the same size as `in1`, centered with respect to the 'full' output. axes : int or array_like of ints or None, optional Axes over which to compute the convolution. The default is over all axes.

Returns ------- out : array An N-dimensional array containing a subset of the discrete linear convolution of `in1` with `in2`.

See Also -------- convolve : Uses the direct convolution or FFT convolution algorithm depending on which is faster. oaconvolve : Uses the overlap-add method to do convolution, which is generally faster when the input arrays are large and significantly different in size.

Examples -------- Autocorrelation of white noise is an impulse.

>>> from scipy import signal >>> sig = np.random.randn(1000) >>> autocorr = signal.fftconvolve(sig, sig::-1, mode='full')

>>> import matplotlib.pyplot as plt >>> fig, (ax_orig, ax_mag) = plt.subplots(2, 1) >>> ax_orig.plot(sig) >>> ax_orig.set_title('White noise') >>> ax_mag.plot(np.arange(-len(sig)+1,len(sig)), autocorr) >>> ax_mag.set_title('Autocorrelation') >>> fig.tight_layout() >>> fig.show()

Gaussian blur implemented using FFT convolution. Notice the dark borders around the image, due to the zero-padding beyond its boundaries. The `convolve2d` function allows for other types of image boundaries, but is far slower.

>>> from scipy import misc >>> face = misc.face(gray=True) >>> kernel = np.outer(signal.gaussian(70, 8), signal.gaussian(70, 8)) >>> blurred = signal.fftconvolve(face, kernel, mode='same')

>>> fig, (ax_orig, ax_kernel, ax_blurred) = plt.subplots(3, 1, ... figsize=(6, 15)) >>> ax_orig.imshow(face, cmap='gray') >>> ax_orig.set_title('Original') >>> ax_orig.set_axis_off() >>> ax_kernel.imshow(kernel, cmap='gray') >>> ax_kernel.set_title('Gaussian kernel') >>> ax_kernel.set_axis_off() >>> ax_blurred.imshow(blurred, cmap='gray') >>> ax_blurred.set_title('Blurred') >>> ax_blurred.set_axis_off() >>> fig.show()

val filtfilt : ?axis:int -> ?padtype:[ `S of string | `None ] -> ?padlen:int -> ?method_:string -> ?irlen:int -> b:[> `Ndarray ] Np.Obj.t -> a:[> `Ndarray ] Np.Obj.t -> x:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Apply a digital filter forward and backward to a signal.

This function applies a linear digital filter twice, once forward and once backwards. The combined filter has zero phase and a filter order twice that of the original.

The function provides options for handling the edges of the signal.

The function `sosfiltfilt` (and filter design using ``output='sos'``) should be preferred over `filtfilt` for most filtering tasks, as second-order sections have fewer numerical problems.

Parameters ---------- b : (N,) array_like The numerator coefficient vector of the filter. a : (N,) array_like The denominator coefficient vector of the filter. If ``a0`` is not 1, then both `a` and `b` are normalized by ``a0``. x : array_like The array of data to be filtered. axis : int, optional The axis of `x` to which the filter is applied. Default is -1. padtype : str or None, optional Must be 'odd', 'even', 'constant', or None. This determines the type of extension to use for the padded signal to which the filter is applied. If `padtype` is None, no padding is used. The default is 'odd'. padlen : int or None, optional The number of elements by which to extend `x` at both ends of `axis` before applying the filter. This value must be less than ``x.shapeaxis - 1``. ``padlen=0`` implies no padding. The default value is ``3 * max(len(a), len(b))``. method : str, optional Determines the method for handling the edges of the signal, either 'pad' or 'gust'. When `method` is 'pad', the signal is padded; the type of padding is determined by `padtype` and `padlen`, and `irlen` is ignored. When `method` is 'gust', Gustafsson's method is used, and `padtype` and `padlen` are ignored. irlen : int or None, optional When `method` is 'gust', `irlen` specifies the length of the impulse response of the filter. If `irlen` is None, no part of the impulse response is ignored. For a long signal, specifying `irlen` can significantly improve the performance of the filter.

Returns ------- y : ndarray The filtered output with the same shape as `x`.

See Also -------- sosfiltfilt, lfilter_zi, lfilter, lfiltic, savgol_filter, sosfilt

Notes ----- When `method` is 'pad', the function pads the data along the given axis in one of three ways: odd, even or constant. The odd and even extensions have the corresponding symmetry about the end point of the data. The constant extension extends the data with the values at the end points. On both the forward and backward passes, the initial condition of the filter is found by using `lfilter_zi` and scaling it by the end point of the extended data.

When `method` is 'gust', Gustafsson's method 1_ is used. Initial conditions are chosen for the forward and backward passes so that the forward-backward filter gives the same result as the backward-forward filter.

The option to use Gustaffson's method was added in scipy version 0.16.0.

References ---------- .. 1 F. Gustaffson, 'Determining the initial states in forward-backward filtering', Transactions on Signal Processing, Vol. 46, pp. 988-992, 1996.

Examples -------- The examples will use several functions from `scipy.signal`.

>>> from scipy import signal >>> import matplotlib.pyplot as plt

First we create a one second signal that is the sum of two pure sine waves, with frequencies 5 Hz and 250 Hz, sampled at 2000 Hz.

>>> t = np.linspace(0, 1.0, 2001) >>> xlow = np.sin(2 * np.pi * 5 * t) >>> xhigh = np.sin(2 * np.pi * 250 * t) >>> x = xlow + xhigh

Now create a lowpass Butterworth filter with a cutoff of 0.125 times the Nyquist frequency, or 125 Hz, and apply it to ``x`` with `filtfilt`. The result should be approximately ``xlow``, with no phase shift.

>>> b, a = signal.butter(8, 0.125) >>> y = signal.filtfilt(b, a, x, padlen=150) >>> np.abs(y - xlow).max() 9.1086182074789912e-06

We get a fairly clean result for this artificial example because the odd extension is exact, and with the moderately long padding, the filter's transients have dissipated by the time the actual data is reached. In general, transient effects at the edges are unavoidable.

The following example demonstrates the option ``method='gust'``.

First, create a filter.

>>> b, a = signal.ellip(4, 0.01, 120, 0.125) # Filter to be applied. >>> np.random.seed(123456)

`sig` is a random input signal to be filtered.

>>> n = 60 >>> sig = np.random.randn(n)**3 + 3*np.random.randn(n).cumsum()

Apply `filtfilt` to `sig`, once using the Gustafsson method, and once using padding, and plot the results for comparison.

>>> fgust = signal.filtfilt(b, a, sig, method='gust') >>> fpad = signal.filtfilt(b, a, sig, padlen=50) >>> plt.plot(sig, 'k-', label='input') >>> plt.plot(fgust, 'b-', linewidth=4, label='gust') >>> plt.plot(fpad, 'c-', linewidth=1.5, label='pad') >>> plt.legend(loc='best') >>> plt.show()

The `irlen` argument can be used to improve the performance of Gustafsson's method.

Estimate the impulse response length of the filter.

>>> z, p, k = signal.tf2zpk(b, a) >>> eps = 1e-9 >>> r = np.max(np.abs(p)) >>> approx_impulse_len = int(np.ceil(np.log(eps) / np.log(r))) >>> approx_impulse_len 137

Apply the filter to a longer signal, with and without the `irlen` argument. The difference between `y1` and `y2` is small. For long signals, using `irlen` gives a significant performance improvement.

>>> x = np.random.randn(5000) >>> y1 = signal.filtfilt(b, a, x, method='gust') >>> y2 = signal.filtfilt(b, a, x, method='gust', irlen=approx_impulse_len) >>> print(np.max(np.abs(y1 - y2))) 1.80056858312e-10

val firwin : ?width:float -> ?window: [ `Tuple_of_string_and_parameter_values of Py.Object.t | `S of string ] -> ?pass_zero:[ `Highpass | `Lowpass | `Bandstop | `Bool of bool | `Bandpass ] -> ?scale:float -> ?nyq:float -> ?fs:float -> numtaps:int -> cutoff:[ `F of float | `T1D_array_like of Py.Object.t ] -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

FIR filter design using the window method.

This function computes the coefficients of a finite impulse response filter. The filter will have linear phase; it will be Type I if `numtaps` is odd and Type II if `numtaps` is even.

Type II filters always have zero response at the Nyquist frequency, so a ValueError exception is raised if firwin is called with `numtaps` even and having a passband whose right end is at the Nyquist frequency.

Parameters ---------- numtaps : int Length of the filter (number of coefficients, i.e. the filter order + 1). `numtaps` must be odd if a passband includes the Nyquist frequency. cutoff : float or 1D array_like Cutoff frequency of filter (expressed in the same units as `fs`) OR an array of cutoff frequencies (that is, band edges). In the latter case, the frequencies in `cutoff` should be positive and monotonically increasing between 0 and `fs/2`. The values 0 and `fs/2` must not be included in `cutoff`. width : float or None, optional If `width` is not None, then assume it is the approximate width of the transition region (expressed in the same units as `fs`) for use in Kaiser FIR filter design. In this case, the `window` argument is ignored. window : string or tuple of string and parameter values, optional Desired window to use. See `scipy.signal.get_window` for a list of windows and required parameters. pass_zero : True, False, 'bandpass', 'lowpass', 'highpass', 'bandstop', optional If True, the gain at the frequency 0 (i.e. the 'DC gain') is 1. If False, the DC gain is 0. Can also be a string argument for the desired filter type (equivalent to ``btype`` in IIR design functions).

.. versionadded:: 1.3.0 Support for string arguments. scale : bool, optional Set to True to scale the coefficients so that the frequency response is exactly unity at a certain frequency. That frequency is either:

  • 0 (DC) if the first passband starts at 0 (i.e. pass_zero is True)
  • `fs/2` (the Nyquist frequency) if the first passband ends at `fs/2` (i.e the filter is a single band highpass filter); center of first passband otherwise

nyq : float, optional *Deprecated. Use `fs` instead.* This is the Nyquist frequency. Each frequency in `cutoff` must be between 0 and `nyq`. Default is 1. fs : float, optional The sampling frequency of the signal. Each frequency in `cutoff` must be between 0 and ``fs/2``. Default is 2.

Returns ------- h : (numtaps,) ndarray Coefficients of length `numtaps` FIR filter.

Raises ------ ValueError If any value in `cutoff` is less than or equal to 0 or greater than or equal to ``fs/2``, if the values in `cutoff` are not strictly monotonically increasing, or if `numtaps` is even but a passband includes the Nyquist frequency.

See Also -------- firwin2 firls minimum_phase remez

Examples -------- Low-pass from 0 to f:

>>> from scipy import signal >>> numtaps = 3 >>> f = 0.1 >>> signal.firwin(numtaps, f) array( 0.06799017, 0.86401967, 0.06799017)

Use a specific window function:

>>> signal.firwin(numtaps, f, window='nuttall') array( 3.56607041e-04, 9.99286786e-01, 3.56607041e-04)

High-pass ('stop' from 0 to f):

>>> signal.firwin(numtaps, f, pass_zero=False) array(-0.00859313, 0.98281375, -0.00859313)

Band-pass:

>>> f1, f2 = 0.1, 0.2 >>> signal.firwin(numtaps, f1, f2, pass_zero=False) array( 0.06301614, 0.88770441, 0.06301614)

Band-stop:

>>> signal.firwin(numtaps, f1, f2) array(-0.00801395, 1.0160279 , -0.00801395)

Multi-band (passbands are 0, f1, f2, f3 and f4, 1):

>>> f3, f4 = 0.3, 0.4 >>> signal.firwin(numtaps, f1, f2, f3, f4) array(-0.01376344, 1.02752689, -0.01376344)

Multi-band (passbands are f1, f2 and f3,f4):

>>> signal.firwin(numtaps, f1, f2, f3, f4, pass_zero=False) array( 0.04890915, 0.91284326, 0.04890915)

val gcd : x:Py.Object.t -> y:Py.Object.t -> unit -> Py.Object.t

greatest common divisor of x and y

val get_window : ?fftbins:bool -> window:[ `Tuple of Py.Object.t | `S of string | `F of float ] -> nx:int -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Return a window of a given length and type.

Parameters ---------- window : string, float, or tuple The type of window to create. See below for more details. Nx : int The number of samples in the window. fftbins : bool, optional If True (default), create a 'periodic' window, ready to use with `ifftshift` and be multiplied by the result of an FFT (see also :func:`~scipy.fft.fftfreq`). If False, create a 'symmetric' window, for use in filter design.

Returns ------- get_window : ndarray Returns a window of length `Nx` and type `window`

Notes ----- Window types:

  • `~scipy.signal.windows.boxcar`
  • `~scipy.signal.windows.triang`
  • `~scipy.signal.windows.blackman`
  • `~scipy.signal.windows.hamming`
  • `~scipy.signal.windows.hann`
  • `~scipy.signal.windows.bartlett`
  • `~scipy.signal.windows.flattop`
  • `~scipy.signal.windows.parzen`
  • `~scipy.signal.windows.bohman`
  • `~scipy.signal.windows.blackmanharris`
  • `~scipy.signal.windows.nuttall`
  • `~scipy.signal.windows.barthann`
  • `~scipy.signal.windows.kaiser` (needs beta)
  • `~scipy.signal.windows.gaussian` (needs standard deviation)
  • `~scipy.signal.windows.general_gaussian` (needs power, width)
  • `~scipy.signal.windows.slepian` (needs width)
  • `~scipy.signal.windows.dpss` (needs normalized half-bandwidth)
  • `~scipy.signal.windows.chebwin` (needs attenuation)
  • `~scipy.signal.windows.exponential` (needs decay scale)
  • `~scipy.signal.windows.tukey` (needs taper fraction)

If the window requires no parameters, then `window` can be a string.

If the window requires parameters, then `window` must be a tuple with the first argument the string name of the window, and the next arguments the needed parameters.

If `window` is a floating point number, it is interpreted as the beta parameter of the `~scipy.signal.windows.kaiser` window.

Each of the window types listed above is also the name of a function that can be called directly to create a window of that type.

Examples -------- >>> from scipy import signal >>> signal.get_window('triang', 7) array( 0.125, 0.375, 0.625, 0.875, 0.875, 0.625, 0.375) >>> signal.get_window(('kaiser', 4.0), 9) array( 0.08848053, 0.29425961, 0.56437221, 0.82160913, 0.97885093, 0.97885093, 0.82160913, 0.56437221, 0.29425961) >>> signal.get_window(4.0, 9) array( 0.08848053, 0.29425961, 0.56437221, 0.82160913, 0.97885093, 0.97885093, 0.82160913, 0.56437221, 0.29425961)

val hilbert : ?n:int -> ?axis:int -> x:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Compute the analytic signal, using the Hilbert transform.

The transformation is done along the last axis by default.

Parameters ---------- x : array_like Signal data. Must be real. N : int, optional Number of Fourier components. Default: ``x.shapeaxis`` axis : int, optional Axis along which to do the transformation. Default: -1.

Returns ------- xa : ndarray Analytic signal of `x`, of each 1-D array along `axis`

Notes ----- The analytic signal ``x_a(t)`` of signal ``x(t)`` is:

.. math:: x_a = F^

1

}

(F(x) 2U) = x + i y

where `F` is the Fourier transform, `U` the unit step function, and `y` the Hilbert transform of `x`. 1_

In other words, the negative half of the frequency spectrum is zeroed out, turning the real-valued signal into a complex signal. The Hilbert transformed signal can be obtained from ``np.imag(hilbert(x))``, and the original signal from ``np.real(hilbert(x))``.

Examples --------- In this example we use the Hilbert transform to determine the amplitude envelope and instantaneous frequency of an amplitude-modulated signal.

>>> import numpy as np >>> import matplotlib.pyplot as plt >>> from scipy.signal import hilbert, chirp

>>> duration = 1.0 >>> fs = 400.0 >>> samples = int(fs*duration) >>> t = np.arange(samples) / fs

We create a chirp of which the frequency increases from 20 Hz to 100 Hz and apply an amplitude modulation.

>>> signal = chirp(t, 20.0, t-1, 100.0) >>> signal *= (1.0 + 0.5 * np.sin(2.0*np.pi*3.0*t) )

The amplitude envelope is given by magnitude of the analytic signal. The instantaneous frequency can be obtained by differentiating the instantaneous phase in respect to time. The instantaneous phase corresponds to the phase angle of the analytic signal.

>>> analytic_signal = hilbert(signal) >>> amplitude_envelope = np.abs(analytic_signal) >>> instantaneous_phase = np.unwrap(np.angle(analytic_signal)) >>> instantaneous_frequency = (np.diff(instantaneous_phase) / ... (2.0*np.pi) * fs)

>>> fig = plt.figure() >>> ax0 = fig.add_subplot(211) >>> ax0.plot(t, signal, label='signal') >>> ax0.plot(t, amplitude_envelope, label='envelope') >>> ax0.set_xlabel('time in seconds') >>> ax0.legend() >>> ax1 = fig.add_subplot(212) >>> ax1.plot(t1:, instantaneous_frequency) >>> ax1.set_xlabel('time in seconds') >>> ax1.set_ylim(0.0, 120.0)

References ---------- .. 1 Wikipedia, 'Analytic signal'. https://en.wikipedia.org/wiki/Analytic_signal .. 2 Leon Cohen, 'Time-Frequency Analysis', 1995. Chapter 2. .. 3 Alan V. Oppenheim, Ronald W. Schafer. Discrete-Time Signal Processing, Third Edition, 2009. Chapter 12. ISBN 13: 978-1292-02572-8

val hilbert2 : ?n:[ `Tuple_of_two_ints of Py.Object.t | `I of int ] -> x:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Compute the '2-D' analytic signal of `x`

Parameters ---------- x : array_like 2-D signal data. N : int or tuple of two ints, optional Number of Fourier components. Default is ``x.shape``

Returns ------- xa : ndarray Analytic signal of `x` taken along axes (0,1).

References ---------- .. 1 Wikipedia, 'Analytic signal', https://en.wikipedia.org/wiki/Analytic_signal

val invres : ?tol:float -> ?rtype:[ `Avg | `Min | `Max ] -> r:[> `Ndarray ] Np.Obj.t -> p:[> `Ndarray ] Np.Obj.t -> k:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Compute b(s) and a(s) from partial fraction expansion.

If `M` is the degree of numerator `b` and `N` the degree of denominator `a`::

b(s) b0 s**(M) + b1 s**(M-1) + ... + bM H(s) = ------ = ------------------------------------------ a(s) a0 s**(N) + a1 s**(N-1) + ... + aN

then the partial-fraction expansion H(s) is defined as::

r0 r1 r-1 = -------- + -------- + ... + --------- + k(s) (s-p0) (s-p1) (s-p-1)

If there are any repeated roots (closer together than `tol`), then H(s) has terms like::

ri ri+1 ri+n-1 -------- + ----------- + ... + ----------- (s-pi) (s-pi)**2 (s-pi)**n

This function is used for polynomials in positive powers of s or z, such as analog filters or digital filters in controls engineering. For negative powers of z (typical for digital filters in DSP), use `invresz`.

Parameters ---------- r : array_like Residues corresponding to the poles. For repeated poles, the residues must be ordered to correspond to ascending by power fractions. p : array_like Poles. Equal poles must be adjacent. k : array_like Coefficients of the direct polynomial term. tol : float, optional The tolerance for two roots to be considered equal in terms of the distance between them. Default is 1e-3. See `unique_roots` for further details. rtype : 'avg', 'min', 'max', optional Method for computing a root to represent a group of identical roots. Default is 'avg'. See `unique_roots` for further details.

Returns ------- b : ndarray Numerator polynomial coefficients. a : ndarray Denominator polynomial coefficients.

See Also -------- residue, invresz, unique_roots

val invresz : ?tol:float -> ?rtype:[ `Avg | `Min | `Max ] -> r:[> `Ndarray ] Np.Obj.t -> p:[> `Ndarray ] Np.Obj.t -> k:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Compute b(z) and a(z) from partial fraction expansion.

If `M` is the degree of numerator `b` and `N` the degree of denominator `a`::

b(z) b0 + b1 z**(-1) + ... + bM z**(-M) H(z) = ------ = ------------------------------------------ a(z) a0 + a1 z**(-1) + ... + aN z**(-N)

then the partial-fraction expansion H(z) is defined as::

r0 r-1 = --------------- + ... + ---------------- + k0 + k1z**(-1) ... (1-p0z**(-1)) (1-p-1z**(-1))

If there are any repeated roots (closer than `tol`), then the partial fraction expansion has terms like::

ri ri+1 ri+n-1 -------------- + ------------------ + ... + ------------------ (1-piz**(-1)) (1-piz**(-1))**2 (1-piz**(-1))**n

This function is used for polynomials in negative powers of z, such as digital filters in DSP. For positive powers, use `invres`.

Parameters ---------- r : array_like Residues corresponding to the poles. For repeated poles, the residues must be ordered to correspond to ascending by power fractions. p : array_like Poles. Equal poles must be adjacent. k : array_like Coefficients of the direct polynomial term. tol : float, optional The tolerance for two roots to be considered equal in terms of the distance between them. Default is 1e-3. See `unique_roots` for further details. rtype : 'avg', 'min', 'max', optional Method for computing a root to represent a group of identical roots. Default is 'avg'. See `unique_roots` for further details.

Returns ------- b : ndarray Numerator polynomial coefficients. a : ndarray Denominator polynomial coefficients.

See Also -------- residuez, unique_roots, invres

val lambertw : ?k:int -> ?tol:float -> z:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

lambertw(z, k=0, tol=1e-8)

Lambert W function.

The Lambert W function `W(z)` is defined as the inverse function of ``w * exp(w)``. In other words, the value of ``W(z)`` is such that ``z = W(z) * exp(W(z))`` for any complex number ``z``.

The Lambert W function is a multivalued function with infinitely many branches. Each branch gives a separate solution of the equation ``z = w exp(w)``. Here, the branches are indexed by the integer `k`.

Parameters ---------- z : array_like Input argument. k : int, optional Branch index. tol : float, optional Evaluation tolerance.

Returns ------- w : array `w` will have the same shape as `z`.

Notes ----- All branches are supported by `lambertw`:

* ``lambertw(z)`` gives the principal solution (branch 0) * ``lambertw(z, k)`` gives the solution on branch `k`

The Lambert W function has two partially real branches: the principal branch (`k = 0`) is real for real ``z > -1/e``, and the ``k = -1`` branch is real for ``-1/e < z < 0``. All branches except ``k = 0`` have a logarithmic singularity at ``z = 0``.

**Possible issues**

The evaluation can become inaccurate very close to the branch point at ``-1/e``. In some corner cases, `lambertw` might currently fail to converge, or can end up on the wrong branch.

**Algorithm**

Halley's iteration is used to invert ``w * exp(w)``, using a first-order asymptotic approximation (O(log(w)) or `O(w)`) as the initial estimate.

The definition, implementation and choice of branches is based on 2_.

See Also -------- wrightomega : the Wright Omega function

References ---------- .. 1 https://en.wikipedia.org/wiki/Lambert_W_function .. 2 Corless et al, 'On the Lambert W function', Adv. Comp. Math. 5 (1996) 329-359. http://www.apmaths.uwo.ca/~djeffrey/Offprints/W-adv-cm.pdf

Examples -------- The Lambert W function is the inverse of ``w exp(w)``:

>>> from scipy.special import lambertw >>> w = lambertw(1) >>> w (0.56714329040978384+0j) >>> w * np.exp(w) (1.0+0j)

Any branch gives a valid inverse:

>>> w = lambertw(1, k=3) >>> w (-2.8535817554090377+17.113535539412148j) >>> w*np.exp(w) (1.0000000000000002+1.609823385706477e-15j)

**Applications to equation-solving**

The Lambert W function may be used to solve various kinds of equations, such as finding the value of the infinite power tower :math:`z^z^{z^{\ldots

}

}

`:

>>> def tower(z, n): ... if n == 0: ... return z ... return z ** tower(z, n-1) ... >>> tower(0.5, 100) 0.641185744504986 >>> -lambertw(-np.log(0.5)) / np.log(0.5) (0.64118574450498589+0j)

val lfilter : ?axis:int -> ?zi:[> `Ndarray ] Np.Obj.t -> b:[> `Ndarray ] Np.Obj.t -> a:[> `Ndarray ] Np.Obj.t -> x:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Filter data along one-dimension with an IIR or FIR filter.

Filter a data sequence, `x`, using a digital filter. This works for many fundamental data types (including Object type). The filter is a direct form II transposed implementation of the standard difference equation (see Notes).

The function `sosfilt` (and filter design using ``output='sos'``) should be preferred over `lfilter` for most filtering tasks, as second-order sections have fewer numerical problems.

Parameters ---------- b : array_like The numerator coefficient vector in a 1-D sequence. a : array_like The denominator coefficient vector in a 1-D sequence. If ``a0`` is not 1, then both `a` and `b` are normalized by ``a0``. x : array_like An N-dimensional input array. axis : int, optional The axis of the input data array along which to apply the linear filter. The filter is applied to each subarray along this axis. Default is -1. zi : array_like, optional Initial conditions for the filter delays. It is a vector (or array of vectors for an N-dimensional input) of length ``max(len(a), len(b)) - 1``. If `zi` is None or is not given then initial rest is assumed. See `lfiltic` for more information.

Returns ------- y : array The output of the digital filter. zf : array, optional If `zi` is None, this is not returned, otherwise, `zf` holds the final filter delay values.

See Also -------- lfiltic : Construct initial conditions for `lfilter`. lfilter_zi : Compute initial state (steady state of step response) for `lfilter`. filtfilt : A forward-backward filter, to obtain a filter with linear phase. savgol_filter : A Savitzky-Golay filter. sosfilt: Filter data using cascaded second-order sections. sosfiltfilt: A forward-backward filter using second-order sections.

Notes ----- The filter function is implemented as a direct II transposed structure. This means that the filter implements::

a0*yn = b0*xn + b1*xn-1 + ... + bM*xn-M

  • a1*yn-1 - ... - aN*yn-N

where `M` is the degree of the numerator, `N` is the degree of the denominator, and `n` is the sample number. It is implemented using the following difference equations (assuming M = N)::

a0*yn = b0 * xn + d0n-1 d0n = b1 * xn - a1 * yn + d1n-1 d1n = b2 * xn - a2 * yn + d2n-1 ... dN-2n = bN-1*xn - aN-1*yn + dN-1n-1 dN-1n = bN * xn - aN * yn

where `d` are the state variables.

The rational transfer function describing this filter in the z-transform domain is::

-1 -M b0 + b1z + ... + bM z Y(z) = -------------------------------- X(z) -1 -N a0 + a1z + ... + aN z

Examples -------- Generate a noisy signal to be filtered:

>>> from scipy import signal >>> import matplotlib.pyplot as plt >>> t = np.linspace(-1, 1, 201) >>> x = (np.sin(2*np.pi*0.75*t*(1-t) + 2.1) + ... 0.1*np.sin(2*np.pi*1.25*t + 1) + ... 0.18*np.cos(2*np.pi*3.85*t)) >>> xn = x + np.random.randn(len(t)) * 0.08

Create an order 3 lowpass butterworth filter:

>>> b, a = signal.butter(3, 0.05)

Apply the filter to xn. Use lfilter_zi to choose the initial condition of the filter:

>>> zi = signal.lfilter_zi(b, a) >>> z, _ = signal.lfilter(b, a, xn, zi=zi*xn0)

Apply the filter again, to have a result filtered at an order the same as filtfilt:

>>> z2, _ = signal.lfilter(b, a, z, zi=zi*z0)

Use filtfilt to apply the filter:

>>> y = signal.filtfilt(b, a, xn)

Plot the original signal and the various filtered versions:

>>> plt.figure >>> plt.plot(t, xn, 'b', alpha=0.75) >>> plt.plot(t, z, 'r--', t, z2, 'r', t, y, 'k') >>> plt.legend(('noisy signal', 'lfilter, once', 'lfilter, twice', ... 'filtfilt'), loc='best') >>> plt.grid(True) >>> plt.show()

val lfilter_zi : b:Py.Object.t -> a:Py.Object.t -> unit -> Py.Object.t

Construct initial conditions for lfilter for step response steady-state.

Compute an initial state `zi` for the `lfilter` function that corresponds to the steady state of the step response.

A typical use of this function is to set the initial state so that the output of the filter starts at the same value as the first element of the signal to be filtered.

Parameters ---------- b, a : array_like (1-D) The IIR filter coefficients. See `lfilter` for more information.

Returns ------- zi : 1-D ndarray The initial state for the filter.

See Also -------- lfilter, lfiltic, filtfilt

Notes ----- A linear filter with order m has a state space representation (A, B, C, D), for which the output y of the filter can be expressed as::

z(n+1) = A*z(n) + B*x(n) y(n) = C*z(n) + D*x(n)

where z(n) is a vector of length m, A has shape (m, m), B has shape (m, 1), C has shape (1, m) and D has shape (1, 1) (assuming x(n) is a scalar). lfilter_zi solves::

zi = A*zi + B

In other words, it finds the initial condition for which the response to an input of all ones is a constant.

Given the filter coefficients `a` and `b`, the state space matrices for the transposed direct form II implementation of the linear filter, which is the implementation used by scipy.signal.lfilter, are::

A = scipy.linalg.companion(a).T B = b1: - a1:*b0

assuming `a0` is 1.0; if `a0` is not 1, `a` and `b` are first divided by a0.

Examples -------- The following code creates a lowpass Butterworth filter. Then it applies that filter to an array whose values are all 1.0; the output is also all 1.0, as expected for a lowpass filter. If the `zi` argument of `lfilter` had not been given, the output would have shown the transient signal.

>>> from numpy import array, ones >>> from scipy.signal import lfilter, lfilter_zi, butter >>> b, a = butter(5, 0.25) >>> zi = lfilter_zi(b, a) >>> y, zo = lfilter(b, a, ones(10), zi=zi) >>> y array(1., 1., 1., 1., 1., 1., 1., 1., 1., 1.)

Another example:

>>> x = array(0.5, 0.5, 0.5, 0.0, 0.0, 0.0, 0.0) >>> y, zf = lfilter(b, a, x, zi=zi*x0) >>> y array( 0.5 , 0.5 , 0.5 , 0.49836039, 0.48610528, 0.44399389, 0.35505241)

Note that the `zi` argument to `lfilter` was computed using `lfilter_zi` and scaled by `x0`. Then the output `y` has no transient until the input drops from 0.5 to 0.0.

val lfiltic : ?x:[> `Ndarray ] Np.Obj.t -> b:[> `Ndarray ] Np.Obj.t -> a:[> `Ndarray ] Np.Obj.t -> y:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Construct initial conditions for lfilter given input and output vectors.

Given a linear filter (b, a) and initial conditions on the output `y` and the input `x`, return the initial conditions on the state vector zi which is used by `lfilter` to generate the output given the input.

Parameters ---------- b : array_like Linear filter term. a : array_like Linear filter term. y : array_like Initial conditions.

If ``N = len(a) - 1``, then ``y = y[-1], y[-2], ..., y[-N]``.

If `y` is too short, it is padded with zeros. x : array_like, optional Initial conditions.

If ``M = len(b) - 1``, then ``x = x[-1], x[-2], ..., x[-M]``.

If `x` is not given, its initial conditions are assumed zero.

If `x` is too short, it is padded with zeros.

Returns ------- zi : ndarray The state vector ``zi = z_0[-1], z_1[-1], ..., z_K-1[-1]``, where ``K = max(M, N)``.

See Also -------- lfilter, lfilter_zi

val medfilt : ?kernel_size:[> `Ndarray ] Np.Obj.t -> volume:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Perform a median filter on an N-dimensional array.

Apply a median filter to the input array using a local window-size given by `kernel_size`. The array will automatically be zero-padded.

Parameters ---------- volume : array_like An N-dimensional input array. kernel_size : array_like, optional A scalar or an N-length list giving the size of the median filter window in each dimension. Elements of `kernel_size` should be odd. If `kernel_size` is a scalar, then this scalar is used as the size in each dimension. Default size is 3 for each dimension.

Returns ------- out : ndarray An array the same size as input containing the median filtered result.

See also -------- scipy.ndimage.median_filter

Notes ------- The more general function `scipy.ndimage.median_filter` has a more efficient implementation of a median filter and therefore runs much faster.

val medfilt2d : ?kernel_size:[> `Ndarray ] Np.Obj.t -> input:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Median filter a 2-dimensional array.

Apply a median filter to the `input` array using a local window-size given by `kernel_size` (must be odd). The array is zero-padded automatically.

Parameters ---------- input : array_like A 2-dimensional input array. kernel_size : array_like, optional A scalar or a list of length 2, giving the size of the median filter window in each dimension. Elements of `kernel_size` should be odd. If `kernel_size` is a scalar, then this scalar is used as the size in each dimension. Default is a kernel of size (3, 3).

Returns ------- out : ndarray An array the same size as input containing the median filtered result.

See also -------- scipy.ndimage.median_filter

Notes ------- The more general function `scipy.ndimage.median_filter` has a more efficient implementation of a median filter and therefore runs much faster.

val oaconvolve : ?mode:[ `Full | `Valid | `Same ] -> ?axes:[ `I of int | `Array_like_of_ints of Py.Object.t ] -> in1:[> `Ndarray ] Np.Obj.t -> in2:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Convolve two N-dimensional arrays using the overlap-add method.

Convolve `in1` and `in2` using the overlap-add method, with the output size determined by the `mode` argument.

This is generally much faster than `convolve` for large arrays (n > ~500), and generally much faster than `fftconvolve` when one array is much larger than the other, but can be slower when only a few output values are needed or when the arrays are very similar in shape, and can only output float arrays (int or object array inputs will be cast to float).

Parameters ---------- in1 : array_like First input. in2 : array_like Second input. Should have the same number of dimensions as `in1`. mode : str 'full', 'valid', 'same', optional A string indicating the size of the output:

``full`` The output is the full discrete linear convolution of the inputs. (Default) ``valid`` The output consists only of those elements that do not rely on the zero-padding. In 'valid' mode, either `in1` or `in2` must be at least as large as the other in every dimension. ``same`` The output is the same size as `in1`, centered with respect to the 'full' output. axes : int or array_like of ints or None, optional Axes over which to compute the convolution. The default is over all axes.

Returns ------- out : array An N-dimensional array containing a subset of the discrete linear convolution of `in1` with `in2`.

See Also -------- convolve : Uses the direct convolution or FFT convolution algorithm depending on which is faster. fftconvolve : An implementation of convolution using FFT.

Notes ----- .. versionadded:: 1.4.0

Examples -------- Convolve a 100,000 sample signal with a 512-sample filter.

>>> from scipy import signal >>> sig = np.random.randn(100000) >>> filt = signal.firwin(512, 0.01) >>> fsig = signal.oaconvolve(sig, filt)

>>> import matplotlib.pyplot as plt >>> fig, (ax_orig, ax_mag) = plt.subplots(2, 1) >>> ax_orig.plot(sig) >>> ax_orig.set_title('White noise') >>> ax_mag.plot(fsig) >>> ax_mag.set_title('Filtered noise') >>> fig.tight_layout() >>> fig.show()

References ---------- .. 1 Wikipedia, 'Overlap-add_method'. https://en.wikipedia.org/wiki/Overlap-add_method .. 2 Richard G. Lyons. Understanding Digital Signal Processing, Third Edition, 2011. Chapter 13.10. ISBN 13: 978-0137-02741-5

val odd_ext : ?axis:int -> x:[> `Ndarray ] Np.Obj.t -> n:int -> unit -> Py.Object.t

Odd extension at the boundaries of an array

Generate a new ndarray by making an odd extension of `x` along an axis.

Parameters ---------- x : ndarray The array to be extended. n : int The number of elements by which to extend `x` at each end of the axis. axis : int, optional The axis along which to extend `x`. Default is -1.

Examples -------- >>> from scipy.signal._arraytools import odd_ext >>> a = np.array([1, 2, 3, 4, 5], [0, 1, 4, 9, 16]) >>> odd_ext(a, 2) array([-1, 0, 1, 2, 3, 4, 5, 6, 7], [-4, -1, 0, 1, 4, 9, 16, 23, 28])

Odd extension is a '180 degree rotation' at the endpoints of the original array:

>>> t = np.linspace(0, 1.5, 100) >>> a = 0.9 * np.sin(2 * np.pi * t**2) >>> b = odd_ext(a, 40) >>> import matplotlib.pyplot as plt >>> plt.plot(arange(-40, 140), b, 'b', lw=1, label='odd extension') >>> plt.plot(arange(100), a, 'r', lw=2, label='original') >>> plt.legend(loc='best') >>> plt.show()

val order_filter : a:[> `Ndarray ] Np.Obj.t -> domain:[> `Ndarray ] Np.Obj.t -> rank:int -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Perform an order filter on an N-dimensional array.

Perform an order filter on the array in. The domain argument acts as a mask centered over each pixel. The non-zero elements of domain are used to select elements surrounding each input pixel which are placed in a list. The list is sorted, and the output for that pixel is the element corresponding to rank in the sorted list.

Parameters ---------- a : ndarray The N-dimensional input array. domain : array_like A mask array with the same number of dimensions as `a`. Each dimension should have an odd number of elements. rank : int A non-negative integer which selects the element from the sorted list (0 corresponds to the smallest element, 1 is the next smallest element, etc.).

Returns ------- out : ndarray The results of the order filter in an array with the same shape as `a`.

Examples -------- >>> from scipy import signal >>> x = np.arange(25).reshape(5, 5) >>> domain = np.identity(3) >>> x array([ 0, 1, 2, 3, 4], [ 5, 6, 7, 8, 9], [10, 11, 12, 13, 14], [15, 16, 17, 18, 19], [20, 21, 22, 23, 24]) >>> signal.order_filter(x, domain, 0) array([ 0., 0., 0., 0., 0.], [ 0., 0., 1., 2., 0.], [ 0., 5., 6., 7., 0.], [ 0., 10., 11., 12., 0.], [ 0., 0., 0., 0., 0.]) >>> signal.order_filter(x, domain, 2) array([ 6., 7., 8., 9., 4.], [ 11., 12., 13., 14., 9.], [ 16., 17., 18., 19., 14.], [ 21., 22., 23., 24., 19.], [ 20., 21., 22., 23., 24.])

val resample : ?t:[> `Ndarray ] Np.Obj.t -> ?axis:int -> ?window: [ `Tuple of Py.Object.t | `S of string | `F of float | `Ndarray of [> `Ndarray ] Np.Obj.t | `Callable of Py.Object.t ] -> x:[> `Ndarray ] Np.Obj.t -> num:int -> unit -> Py.Object.t

Resample `x` to `num` samples using Fourier method along the given axis.

The resampled signal starts at the same value as `x` but is sampled with a spacing of ``len(x) / num * (spacing of x)``. Because a Fourier method is used, the signal is assumed to be periodic.

Parameters ---------- x : array_like The data to be resampled. num : int The number of samples in the resampled signal. t : array_like, optional If `t` is given, it is assumed to be the equally spaced sample positions associated with the signal data in `x`. axis : int, optional The axis of `x` that is resampled. Default is 0. window : array_like, callable, string, float, or tuple, optional Specifies the window applied to the signal in the Fourier domain. See below for details.

Returns ------- resampled_x or (resampled_x, resampled_t) Either the resampled array, or, if `t` was given, a tuple containing the resampled array and the corresponding resampled positions.

See Also -------- decimate : Downsample the signal after applying an FIR or IIR filter. resample_poly : Resample using polyphase filtering and an FIR filter.

Notes ----- The argument `window` controls a Fourier-domain window that tapers the Fourier spectrum before zero-padding to alleviate ringing in the resampled values for sampled signals you didn't intend to be interpreted as band-limited.

If `window` is a function, then it is called with a vector of inputs indicating the frequency bins (i.e. fftfreq(x.shapeaxis) ).

If `window` is an array of the same length as `x.shapeaxis` it is assumed to be the window to be applied directly in the Fourier domain (with dc and low-frequency first).

For any other type of `window`, the function `scipy.signal.get_window` is called to generate the window.

The first sample of the returned vector is the same as the first sample of the input vector. The spacing between samples is changed from ``dx`` to ``dx * len(x) / num``.

If `t` is not None, then it is used solely to calculate the resampled positions `resampled_t`

As noted, `resample` uses FFT transformations, which can be very slow if the number of input or output samples is large and prime; see `scipy.fft.fft`.

Examples -------- Note that the end of the resampled data rises to meet the first sample of the next cycle:

>>> from scipy import signal

>>> x = np.linspace(0, 10, 20, endpoint=False) >>> y = np.cos(-x**2/6.0) >>> f = signal.resample(y, 100) >>> xnew = np.linspace(0, 10, 100, endpoint=False)

>>> import matplotlib.pyplot as plt >>> plt.plot(x, y, 'go-', xnew, f, '.-', 10, y0, 'ro') >>> plt.legend('data', 'resampled', loc='best') >>> plt.show()

val resample_poly : ?axis:int -> ?window: [ `Ndarray of [> `Ndarray ] Np.Obj.t | `Tuple of Py.Object.t | `S of string ] -> ?padtype:string -> ?cval:float -> x:[> `Ndarray ] Np.Obj.t -> up:int -> down:int -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Resample `x` along the given axis using polyphase filtering.

The signal `x` is upsampled by the factor `up`, a zero-phase low-pass FIR filter is applied, and then it is downsampled by the factor `down`. The resulting sample rate is ``up / down`` times the original sample rate. By default, values beyond the boundary of the signal are assumed to be zero during the filtering step.

Parameters ---------- x : array_like The data to be resampled. up : int The upsampling factor. down : int The downsampling factor. axis : int, optional The axis of `x` that is resampled. Default is 0. window : string, tuple, or array_like, optional Desired window to use to design the low-pass filter, or the FIR filter coefficients to employ. See below for details. padtype : string, optional `constant`, `line`, `mean`, `median`, `maximum`, `minimum` or any of the other signal extension modes supported by `scipy.signal.upfirdn`. Changes assumptions on values beyond the boundary. If `constant`, assumed to be `cval` (default zero). If `line` assumed to continue a linear trend defined by the first and last points. `mean`, `median`, `maximum` and `minimum` work as in `np.pad` and assume that the values beyond the boundary are the mean, median, maximum or minimum respectively of the array along the axis.

.. versionadded:: 1.4.0 cval : float, optional Value to use if `padtype='constant'`. Default is zero.

.. versionadded:: 1.4.0

Returns ------- resampled_x : array The resampled array.

See Also -------- decimate : Downsample the signal after applying an FIR or IIR filter. resample : Resample up or down using the FFT method.

Notes ----- This polyphase method will likely be faster than the Fourier method in `scipy.signal.resample` when the number of samples is large and prime, or when the number of samples is large and `up` and `down` share a large greatest common denominator. The length of the FIR filter used will depend on ``max(up, down) // gcd(up, down)``, and the number of operations during polyphase filtering will depend on the filter length and `down` (see `scipy.signal.upfirdn` for details).

The argument `window` specifies the FIR low-pass filter design.

If `window` is an array_like it is assumed to be the FIR filter coefficients. Note that the FIR filter is applied after the upsampling step, so it should be designed to operate on a signal at a sampling frequency higher than the original by a factor of `up//gcd(up, down)`. This function's output will be centered with respect to this array, so it is best to pass a symmetric filter with an odd number of samples if, as is usually the case, a zero-phase filter is desired.

For any other type of `window`, the functions `scipy.signal.get_window` and `scipy.signal.firwin` are called to generate the appropriate filter coefficients.

The first sample of the returned vector is the same as the first sample of the input vector. The spacing between samples is changed from ``dx`` to ``dx * down / float(up)``.

Examples -------- By default, the end of the resampled data rises to meet the first sample of the next cycle for the FFT method, and gets closer to zero for the polyphase method:

>>> from scipy import signal

>>> x = np.linspace(0, 10, 20, endpoint=False) >>> y = np.cos(-x**2/6.0) >>> f_fft = signal.resample(y, 100) >>> f_poly = signal.resample_poly(y, 100, 20) >>> xnew = np.linspace(0, 10, 100, endpoint=False)

>>> import matplotlib.pyplot as plt >>> plt.plot(xnew, f_fft, 'b.-', xnew, f_poly, 'r.-') >>> plt.plot(x, y, 'ko-') >>> plt.plot(10, y0, 'bo', 10, 0., 'ro') # boundaries >>> plt.legend('resample', 'resamp_poly', 'data', loc='best') >>> plt.show()

This default behaviour can be changed by using the padtype option:

>>> import numpy as np >>> from scipy import signal

>>> N = 5 >>> x = np.linspace(0, 1, N, endpoint=False) >>> y = 2 + x**2 - 1.7*np.sin(x) + .2*np.cos(11*x) >>> y2 = 1 + x**3 + 0.1*np.sin(x) + .1*np.cos(11*x) >>> Y = np.stack(y, y2, axis=-1) >>> up = 4 >>> xr = np.linspace(0, 1, N*up, endpoint=False)

>>> y2 = signal.resample_poly(Y, up, 1, padtype='constant') >>> y3 = signal.resample_poly(Y, up, 1, padtype='mean') >>> y4 = signal.resample_poly(Y, up, 1, padtype='line')

>>> import matplotlib.pyplot as plt >>> for i in 0,1: ... plt.figure() ... plt.plot(xr, y4:,i, 'g.', label='line') ... plt.plot(xr, y3:,i, 'y.', label='mean') ... plt.plot(xr, y2:,i, 'r.', label='constant') ... plt.plot(x, Y:,i, 'k-') ... plt.legend() >>> plt.show()

val residue : ?tol:float -> ?rtype:[ `Avg | `Min | `Max ] -> b:[> `Ndarray ] Np.Obj.t -> a:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Compute partial-fraction expansion of b(s) / a(s).

If `M` is the degree of numerator `b` and `N` the degree of denominator `a`::

b(s) b0 s**(M) + b1 s**(M-1) + ... + bM H(s) = ------ = ------------------------------------------ a(s) a0 s**(N) + a1 s**(N-1) + ... + aN

then the partial-fraction expansion H(s) is defined as::

r0 r1 r-1 = -------- + -------- + ... + --------- + k(s) (s-p0) (s-p1) (s-p-1)

If there are any repeated roots (closer together than `tol`), then H(s) has terms like::

ri ri+1 ri+n-1 -------- + ----------- + ... + ----------- (s-pi) (s-pi)**2 (s-pi)**n

This function is used for polynomials in positive powers of s or z, such as analog filters or digital filters in controls engineering. For negative powers of z (typical for digital filters in DSP), use `residuez`.

See Notes for details about the algorithm.

Parameters ---------- b : array_like Numerator polynomial coefficients. a : array_like Denominator polynomial coefficients. tol : float, optional The tolerance for two roots to be considered equal in terms of the distance between them. Default is 1e-3. See `unique_roots` for further details. rtype : 'avg', 'min', 'max', optional Method for computing a root to represent a group of identical roots. Default is 'avg'. See `unique_roots` for further details.

Returns ------- r : ndarray Residues corresponding to the poles. For repeated poles, the residues are ordered to correspond to ascending by power fractions. p : ndarray Poles ordered by magnitude in ascending order. k : ndarray Coefficients of the direct polynomial term.

See Also -------- invres, residuez, numpy.poly, unique_roots

Notes ----- The 'deflation through subtraction' algorithm is used for computations --- method 6 in 1_.

The form of partial fraction expansion depends on poles multiplicity in the exact mathematical sense. However there is no way to exactly determine multiplicity of roots of a polynomial in numerical computing. Thus you should think of the result of `residue` with given `tol` as partial fraction expansion computed for the denominator composed of the computed poles with empirically determined multiplicity. The choice of `tol` can drastically change the result if there are close poles.

References ---------- .. 1 J. F. Mahoney, B. D. Sivazlian, 'Partial fractions expansion: a review of computational methodology and efficiency', Journal of Computational and Applied Mathematics, Vol. 9, 1983.

val residuez : ?tol:float -> ?rtype:[ `Avg | `Min | `Max ] -> b:[> `Ndarray ] Np.Obj.t -> a:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Compute partial-fraction expansion of b(z) / a(z).

If `M` is the degree of numerator `b` and `N` the degree of denominator `a`::

b(z) b0 + b1 z**(-1) + ... + bM z**(-M) H(z) = ------ = ------------------------------------------ a(z) a0 + a1 z**(-1) + ... + aN z**(-N)

then the partial-fraction expansion H(z) is defined as::

r0 r-1 = --------------- + ... + ---------------- + k0 + k1z**(-1) ... (1-p0z**(-1)) (1-p-1z**(-1))

If there are any repeated roots (closer than `tol`), then the partial fraction expansion has terms like::

ri ri+1 ri+n-1 -------------- + ------------------ + ... + ------------------ (1-piz**(-1)) (1-piz**(-1))**2 (1-piz**(-1))**n

This function is used for polynomials in negative powers of z, such as digital filters in DSP. For positive powers, use `residue`.

See Notes of `residue` for details about the algorithm.

Parameters ---------- b : array_like Numerator polynomial coefficients. a : array_like Denominator polynomial coefficients. tol : float, optional The tolerance for two roots to be considered equal in terms of the distance between them. Default is 1e-3. See `unique_roots` for further details. rtype : 'avg', 'min', 'max', optional Method for computing a root to represent a group of identical roots. Default is 'avg'. See `unique_roots` for further details.

Returns ------- r : ndarray Residues corresponding to the poles. For repeated poles, the residues are ordered to correspond to ascending by power fractions. p : ndarray Poles ordered by magnitude in ascending order. k : ndarray Coefficients of the direct polynomial term.

See Also -------- invresz, residue, unique_roots

val sosfilt : ?axis:int -> ?zi:[> `Ndarray ] Np.Obj.t -> sos:[> `Ndarray ] Np.Obj.t -> x:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Filter data along one dimension using cascaded second-order sections.

Filter a data sequence, `x`, using a digital IIR filter defined by `sos`.

Parameters ---------- sos : array_like Array of second-order filter coefficients, must have shape ``(n_sections, 6)``. Each row corresponds to a second-order section, with the first three columns providing the numerator coefficients and the last three providing the denominator coefficients. x : array_like An N-dimensional input array. axis : int, optional The axis of the input data array along which to apply the linear filter. The filter is applied to each subarray along this axis. Default is -1. zi : array_like, optional Initial conditions for the cascaded filter delays. It is a (at least 2D) vector of shape ``(n_sections, ..., 2, ...)``, where ``..., 2, ...`` denotes the shape of `x`, but with ``x.shapeaxis`` replaced by 2. If `zi` is None or is not given then initial rest (i.e. all zeros) is assumed. Note that these initial conditions are *not* the same as the initial conditions given by `lfiltic` or `lfilter_zi`.

Returns ------- y : ndarray The output of the digital filter. zf : ndarray, optional If `zi` is None, this is not returned, otherwise, `zf` holds the final filter delay values.

See Also -------- zpk2sos, sos2zpk, sosfilt_zi, sosfiltfilt, sosfreqz

Notes ----- The filter function is implemented as a series of second-order filters with direct-form II transposed structure. It is designed to minimize numerical precision errors for high-order filters.

.. versionadded:: 0.16.0

Examples -------- Plot a 13th-order filter's impulse response using both `lfilter` and `sosfilt`, showing the instability that results from trying to do a 13th-order filter in a single stage (the numerical error pushes some poles outside of the unit circle):

>>> import matplotlib.pyplot as plt >>> from scipy import signal >>> b, a = signal.ellip(13, 0.009, 80, 0.05, output='ba') >>> sos = signal.ellip(13, 0.009, 80, 0.05, output='sos') >>> x = signal.unit_impulse(700) >>> y_tf = signal.lfilter(b, a, x) >>> y_sos = signal.sosfilt(sos, x) >>> plt.plot(y_tf, 'r', label='TF') >>> plt.plot(y_sos, 'k', label='SOS') >>> plt.legend(loc='best') >>> plt.show()

val sosfilt_zi : [> `Ndarray ] Np.Obj.t -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Construct initial conditions for sosfilt for step response steady-state.

Compute an initial state `zi` for the `sosfilt` function that corresponds to the steady state of the step response.

A typical use of this function is to set the initial state so that the output of the filter starts at the same value as the first element of the signal to be filtered.

Parameters ---------- sos : array_like Array of second-order filter coefficients, must have shape ``(n_sections, 6)``. See `sosfilt` for the SOS filter format specification.

Returns ------- zi : ndarray Initial conditions suitable for use with ``sosfilt``, shape ``(n_sections, 2)``.

See Also -------- sosfilt, zpk2sos

Notes ----- .. versionadded:: 0.16.0

Examples -------- Filter a rectangular pulse that begins at time 0, with and without the use of the `zi` argument of `scipy.signal.sosfilt`.

>>> from scipy import signal >>> import matplotlib.pyplot as plt

>>> sos = signal.butter(9, 0.125, output='sos') >>> zi = signal.sosfilt_zi(sos) >>> x = (np.arange(250) < 100).astype(int) >>> f1 = signal.sosfilt(sos, x) >>> f2, zo = signal.sosfilt(sos, x, zi=zi)

>>> plt.plot(x, 'k--', label='x') >>> plt.plot(f1, 'b', alpha=0.5, linewidth=2, label='filtered') >>> plt.plot(f2, 'g', alpha=0.25, linewidth=4, label='filtered with zi') >>> plt.legend(loc='best') >>> plt.show()

val sosfiltfilt : ?axis:int -> ?padtype:[ `S of string | `None ] -> ?padlen:int -> sos:[> `Ndarray ] Np.Obj.t -> x:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

A forward-backward digital filter using cascaded second-order sections.

See `filtfilt` for more complete information about this method.

Parameters ---------- sos : array_like Array of second-order filter coefficients, must have shape ``(n_sections, 6)``. Each row corresponds to a second-order section, with the first three columns providing the numerator coefficients and the last three providing the denominator coefficients. x : array_like The array of data to be filtered. axis : int, optional The axis of `x` to which the filter is applied. Default is -1. padtype : str or None, optional Must be 'odd', 'even', 'constant', or None. This determines the type of extension to use for the padded signal to which the filter is applied. If `padtype` is None, no padding is used. The default is 'odd'. padlen : int or None, optional The number of elements by which to extend `x` at both ends of `axis` before applying the filter. This value must be less than ``x.shapeaxis - 1``. ``padlen=0`` implies no padding. The default value is::

3 * (2 * len(sos) + 1 - min((sos:, 2 == 0).sum(), (sos:, 5 == 0).sum()))

The extra subtraction at the end attempts to compensate for poles and zeros at the origin (e.g. for odd-order filters) to yield equivalent estimates of `padlen` to those of `filtfilt` for second-order section filters built with `scipy.signal` functions.

Returns ------- y : ndarray The filtered output with the same shape as `x`.

See Also -------- filtfilt, sosfilt, sosfilt_zi, sosfreqz

Notes ----- .. versionadded:: 0.18.0

Examples -------- >>> from scipy.signal import sosfiltfilt, butter >>> import matplotlib.pyplot as plt

Create an interesting signal to filter.

>>> n = 201 >>> t = np.linspace(0, 1, n) >>> np.random.seed(123) >>> x = 1 + (t < 0.5) - 0.25*t**2 + 0.05*np.random.randn(n)

Create a lowpass Butterworth filter, and use it to filter `x`.

>>> sos = butter(4, 0.125, output='sos') >>> y = sosfiltfilt(sos, x)

For comparison, apply an 8th order filter using `sosfilt`. The filter is initialized using the mean of the first four values of `x`.

>>> from scipy.signal import sosfilt, sosfilt_zi >>> sos8 = butter(8, 0.125, output='sos') >>> zi = x:4.mean() * sosfilt_zi(sos8) >>> y2, zo = sosfilt(sos8, x, zi=zi)

Plot the results. Note that the phase of `y` matches the input, while `y2` has a significant phase delay.

>>> plt.plot(t, x, alpha=0.5, label='x(t)') >>> plt.plot(t, y, label='y(t)') >>> plt.plot(t, y2, label='y2(t)') >>> plt.legend(framealpha=1, shadow=True) >>> plt.grid(alpha=0.25) >>> plt.xlabel('t') >>> plt.show()

val sp_fft : ?kwargs:(string * Py.Object.t) list -> Py.Object.t list -> Py.Object.t

============================================== Discrete Fourier transforms (:mod:`scipy.fft`) ==============================================

.. currentmodule:: scipy.fft

Fast Fourier Transforms (FFTs) ==============================

.. autosummary:: :toctree: generated/

fft - Fast (discrete) Fourier Transform (FFT) ifft - Inverse FFT fft2 - Two dimensional FFT ifft2 - Two dimensional inverse FFT fftn - n-dimensional FFT ifftn - n-dimensional inverse FFT rfft - FFT of strictly real-valued sequence irfft - Inverse of rfft rfft2 - Two dimensional FFT of real sequence irfft2 - Inverse of rfft2 rfftn - n-dimensional FFT of real sequence irfftn - Inverse of rfftn hfft - FFT of a Hermitian sequence (real spectrum) ihfft - Inverse of hfft hfft2 - Two dimensional FFT of a Hermitian sequence ihfft2 - Inverse of hfft2 hfftn - n-dimensional FFT of a Hermitian sequence ihfftn - Inverse of hfftn

Discrete Sin and Cosine Transforms (DST and DCT) ================================================ .. autosummary:: :toctree: generated/

dct - Discrete cosine transform idct - Inverse discrete cosine transform dctn - n-dimensional Discrete cosine transform idctn - n-dimensional Inverse discrete cosine transform dst - Discrete sine transform idst - Inverse discrete sine transform dstn - n-dimensional Discrete sine transform idstn - n-dimensional Inverse discrete sine transform

Helper functions ================

.. autosummary:: :toctree: generated/

fftshift - Shift the zero-frequency component to the center of the spectrum ifftshift - The inverse of `fftshift` fftfreq - Return the Discrete Fourier Transform sample frequencies rfftfreq - DFT sample frequencies (for usage with rfft, irfft) next_fast_len - Find the optimal length to zero-pad an FFT for speed set_workers - Context manager to set default number of workers get_workers - Get the current default number of workers

Backend control ===============

.. autosummary:: :toctree: generated/

set_backend - Context manager to set the backend within a fixed scope skip_backend - Context manager to skip a backend within a fixed scope set_global_backend - Sets the global fft backend register_backend - Register a backend for permanent use

val unique_roots : ?tol:float -> ?rtype:[ `Max | `Maximum | `Min | `Minimum | `Avg | `Mean ] -> p:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t * [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Determine unique roots and their multiplicities from a list of roots.

Parameters ---------- p : array_like The list of roots. tol : float, optional The tolerance for two roots to be considered equal in terms of the distance between them. Default is 1e-3. Refer to Notes about the details on roots grouping. rtype : 'max', 'maximum', 'min', 'minimum', 'avg', 'mean', optional How to determine the returned root if multiple roots are within `tol` of each other.

  • 'max', 'maximum': pick the maximum of those roots
  • 'min', 'minimum': pick the minimum of those roots
  • 'avg', 'mean': take the average of those roots

When finding minimum or maximum among complex roots they are compared first by the real part and then by the imaginary part.

Returns ------- unique : ndarray The list of unique roots. multiplicity : ndarray The multiplicity of each root.

Notes ----- If we have 3 roots ``a``, ``b`` and ``c``, such that ``a`` is close to ``b`` and ``b`` is close to ``c`` (distance is less than `tol`), then it doesn't necessarily mean that ``a`` is close to ``c``. It means that roots grouping is not unique. In this function we use 'greedy' grouping going through the roots in the order they are given in the input `p`.

This utility function is not specific to roots but can be used for any sequence of values for which uniqueness and multiplicity has to be determined. For a more general routine, see `numpy.unique`.

Examples -------- >>> from scipy import signal >>> vals = 0, 1.3, 1.31, 2.8, 1.25, 2.2, 10.3 >>> uniq, mult = signal.unique_roots(vals, tol=2e-2, rtype='avg')

Check which roots have multiplicity larger than 1:

>>> uniqmult > 1 array( 1.305)

val upfirdn : ?up:int -> ?down:int -> ?axis:int -> ?mode:string -> ?cval:float -> h:[> `Ndarray ] Np.Obj.t -> x:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Upsample, FIR filter, and downsample

Parameters ---------- h : array_like 1-dimensional FIR (finite-impulse response) filter coefficients. x : array_like Input signal array. up : int, optional Upsampling rate. Default is 1. down : int, optional Downsampling rate. Default is 1. axis : int, optional The axis of the input data array along which to apply the linear filter. The filter is applied to each subarray along this axis. Default is -1. mode : str, optional The signal extension mode to use. The set ``'constant', 'symmetric', 'reflect', 'edge', 'wrap'`` correspond to modes provided by `numpy.pad`. ``'smooth'`` implements a smooth extension by extending based on the slope of the last 2 points at each end of the array. ``'antireflect'`` and ``'antisymmetric'`` are anti-symmetric versions of ``'reflect'`` and ``'symmetric'``. The mode `'line'` extends the signal based on a linear trend defined by the first and last points along the ``axis``.

.. versionadded:: 1.4.0 cval : float, optional The constant value to use when ``mode == 'constant'``.

.. versionadded:: 1.4.0

Returns ------- y : ndarray The output signal array. Dimensions will be the same as `x` except for along `axis`, which will change size according to the `h`, `up`, and `down` parameters.

Notes ----- The algorithm is an implementation of the block diagram shown on page 129 of the Vaidyanathan text 1_ (Figure 4.3-8d).

.. 1 P. P. Vaidyanathan, Multirate Systems and Filter Banks, Prentice Hall, 1993.

The direct approach of upsampling by factor of P with zero insertion, FIR filtering of length ``N``, and downsampling by factor of Q is O(N*Q) per output sample. The polyphase implementation used here is O(N/P).

.. versionadded:: 0.18

Examples -------- Simple operations:

>>> from scipy.signal import upfirdn >>> upfirdn(1, 1, 1, 1, 1, 1) # FIR filter array( 1., 2., 3., 2., 1.) >>> upfirdn(1, 1, 2, 3, 3) # upsampling with zeros insertion array( 1., 0., 0., 2., 0., 0., 3., 0., 0.) >>> upfirdn(1, 1, 1, 1, 2, 3, 3) # upsampling with sample-and-hold array( 1., 1., 1., 2., 2., 2., 3., 3., 3.) >>> upfirdn(.5, 1, .5, 1, 1, 1, 2) # linear interpolation array( 0.5, 1. , 1. , 1. , 1. , 1. , 0.5, 0. ) >>> upfirdn(1, np.arange(10), 1, 3) # decimation by 3 array( 0., 3., 6., 9.) >>> upfirdn(.5, 1, .5, np.arange(10), 2, 3) # linear interp, rate 2/3 array( 0. , 1. , 2.5, 4. , 5.5, 7. , 8.5, 0. )

Apply a single filter to multiple signals:

>>> x = np.reshape(np.arange(8), (4, 2)) >>> x array([0, 1], [2, 3], [4, 5], [6, 7])

Apply along the last dimension of ``x``:

>>> h = 1, 1 >>> upfirdn(h, x, 2) array([ 0., 0., 1., 1.], [ 2., 2., 3., 3.], [ 4., 4., 5., 5.], [ 6., 6., 7., 7.])

Apply along the 0th dimension of ``x``:

>>> upfirdn(h, x, 2, axis=0) array([ 0., 1.], [ 0., 1.], [ 2., 3.], [ 2., 3.], [ 4., 5.], [ 4., 5.], [ 6., 7.], [ 6., 7.])

val vectorstrength : events:Py.Object.t -> period:[ `Ndarray of [> `Ndarray ] Np.Obj.t | `F of float ] -> unit -> Py.Object.t * Py.Object.t

Determine the vector strength of the events corresponding to the given period.

The vector strength is a measure of phase synchrony, how well the timing of the events is synchronized to a single period of a periodic signal.

If multiple periods are used, calculate the vector strength of each. This is called the 'resonating vector strength'.

Parameters ---------- events : 1D array_like An array of time points containing the timing of the events. period : float or array_like The period of the signal that the events should synchronize to. The period is in the same units as `events`. It can also be an array of periods, in which case the outputs are arrays of the same length.

Returns ------- strength : float or 1D array The strength of the synchronization. 1.0 is perfect synchronization and 0.0 is no synchronization. If `period` is an array, this is also an array with each element containing the vector strength at the corresponding period. phase : float or array The phase that the events are most strongly synchronized to in radians. If `period` is an array, this is also an array with each element containing the phase for the corresponding period.

References ---------- van Hemmen, JL, Longtin, A, and Vollmayr, AN. Testing resonating vector strength: Auditory system, electric fish, and noise. Chaos 21, 047508 (2011); :doi:`10.1063/1.3670512`. van Hemmen, JL. Vector strength after Goldberg, Brown, and von Mises: biological and mathematical perspectives. Biol Cybern. 2013 Aug;107(4):385-96. :doi:`10.1007/s00422-013-0561-7`. van Hemmen, JL and Vollmayr, AN. Resonating vector strength: what happens when we vary the 'probing' frequency while keeping the spike times fixed. Biol Cybern. 2013 Aug;107(4):491-94. :doi:`10.1007/s00422-013-0560-8`.

val wiener : ?mysize:[ `Ndarray of [> `Ndarray ] Np.Obj.t | `I of int ] -> ?noise:float -> im:[> `Ndarray ] Np.Obj.t -> unit -> [ `ArrayLike | `Ndarray | `Object ] Np.Obj.t

Perform a Wiener filter on an N-dimensional array.

Apply a Wiener filter to the N-dimensional array `im`.

Parameters ---------- im : ndarray An N-dimensional array. mysize : int or array_like, optional A scalar or an N-length list giving the size of the Wiener filter window in each dimension. Elements of mysize should be odd. If mysize is a scalar, then this scalar is used as the size in each dimension. noise : float, optional The noise-power to use. If None, then noise is estimated as the average of the local variance of the input.

Returns ------- out : ndarray Wiener filtered result with the same shape as `im`.

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