# Author: Travis Oliphant # 1999 -- 2002
atleast_1d, atleast_2d, cast, dot, exp, expand_dims, iscomplexobj, mean, ndarray, newaxis, ones, pi, poly, polyadd, polyder, polydiv, polymul, polysub, polyval, product, r_, ravel, real_if_close, reshape, roots, sort, take, transpose, unique, where, zeros, zeros_like)
else: from fractions import gcd
'order_filter', 'medfilt', 'medfilt2d', 'wiener', 'lfilter', 'lfiltic', 'sosfilt', 'deconvolve', 'hilbert', 'hilbert2', 'cmplx_sort', 'unique_roots', 'invres', 'invresz', 'residue', 'residuez', 'resample', 'resample_poly', 'detrend', 'lfilter_zi', 'sosfilt_zi', 'sosfiltfilt', 'choose_conv_method', 'filtfilt', 'decimate', 'vectorstrength']
'symmetric': 1, 'reflect': 4}
try: return _modedict[mode] except KeyError: raise ValueError("Acceptable mode flags are 'valid'," " 'same', or 'full'.")
try: return _boundarydict[boundary] << 2 except KeyError: raise ValueError("Acceptable boundary flags are 'fill', 'circular' " "(or 'wrap'), and 'symmetric' (or 'symm').")
""" If in 'valid' mode, returns whether or not the input arrays need to be swapped depending on whether `shape1` is at least as large as `shape2` in every dimension.
This is important for some of the correlation and convolution implementations in this module, where the larger array input needs to come before the smaller array input when operating in this mode.
Note that if the mode provided is not 'valid', False is immediately returned. """ if mode == 'valid': ok1, ok2 = True, True
for d1, d2 in zip(shape1, shape2): if not d1 >= d2: ok1 = False if not d2 >= d1: ok2 = False
if not (ok1 or ok2): raise ValueError("For 'valid' mode, one must be at least " "as large as the other in every dimension")
return not ok1
return False
r""" 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::
z[k] = (x * y)(k - N + 1) = \sum_{l=0}^{||x||-1}x_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, sig[clock], '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, corr[clock], '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()
""" in1 = asarray(in1) in2 = asarray(in2)
if in1.ndim == in2.ndim == 0: return in1 * in2.conj() elif in1.ndim != in2.ndim: raise ValueError("in1 and in2 should have the same dimensionality")
# Don't use _valfrommode, since correlate should not accept numeric modes try: val = _modedict[mode] except KeyError: raise ValueError("Acceptable mode flags are 'valid'," " 'same', or 'full'.")
# this either calls fftconvolve or this function with method=='direct' if method in ('fft', 'auto'): return convolve(in1, _reverse_and_conj(in2), mode, method)
elif method == 'direct': # fastpath to faster numpy.correlate for 1d inputs when possible if _np_conv_ok(in1, in2, mode): return np.correlate(in1, in2, mode)
# _correlateND is far slower when in2.size > in1.size, so swap them # and then undo the effect afterward if mode == 'full'. Also, it fails # with 'valid' mode if in2 is larger than in1, so swap those, too. # Don't swap inputs for 'same' mode, since shape of in1 matters. swapped_inputs = ((mode == 'full') and (in2.size > in1.size) or _inputs_swap_needed(mode, in1.shape, in2.shape))
if swapped_inputs: in1, in2 = in2, in1
if mode == 'valid': ps = [i - j + 1 for i, j in zip(in1.shape, in2.shape)] out = np.empty(ps, in1.dtype)
z = sigtools._correlateND(in1, in2, out, val)
else: ps = [i + j - 1 for i, j in zip(in1.shape, in2.shape)]
# zero pad input in1zpadded = np.zeros(ps, in1.dtype) sc = [slice(0, i) for i in in1.shape] in1zpadded[sc] = in1.copy()
if mode == 'full': out = np.empty(ps, in1.dtype) elif mode == 'same': out = np.empty(in1.shape, in1.dtype)
z = sigtools._correlateND(in1zpadded, in2, out, val)
if swapped_inputs: # Reverse and conjugate to undo the effect of swapping inputs z = _reverse_and_conj(z)
return z
else: raise ValueError("Acceptable method flags are 'auto'," " 'direct', or 'fft'.")
# Return the center newshape portion of the array. newshape = asarray(newshape) currshape = array(arr.shape) startind = (currshape - newshape) // 2 endind = startind + newshape myslice = [slice(startind[k], endind[k]) for k in range(len(endind))] return arr[tuple(myslice)]
"""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.
Returns ------- out : array An N-dimensional array containing a subset of the discrete linear convolution of `in1` with `in2`.
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()
""" in1 = asarray(in1) in2 = asarray(in2)
if in1.ndim == in2.ndim == 0: # scalar inputs return in1 * in2 elif not in1.ndim == in2.ndim: raise ValueError("in1 and in2 should have the same dimensionality") elif in1.size == 0 or in2.size == 0: # empty arrays return array([])
s1 = array(in1.shape) s2 = array(in2.shape) complex_result = (np.issubdtype(in1.dtype, np.complexfloating) or np.issubdtype(in2.dtype, np.complexfloating)) shape = s1 + s2 - 1
# Check that input sizes are compatible with 'valid' mode if _inputs_swap_needed(mode, s1, s2): # Convolution is commutative; order doesn't have any effect on output in1, s1, in2, s2 = in2, s2, in1, s1
# Speed up FFT by padding to optimal size for FFTPACK fshape = [fftpack.helper.next_fast_len(int(d)) for d in shape] fslice = tuple([slice(0, int(sz)) for sz in shape]) # Pre-1.9 NumPy FFT routines are not threadsafe. For older NumPys, make # sure we only call rfftn/irfftn from one thread at a time. if not complex_result and (_rfft_mt_safe or _rfft_lock.acquire(False)): try: sp1 = np.fft.rfftn(in1, fshape) sp2 = np.fft.rfftn(in2, fshape) ret = (np.fft.irfftn(sp1 * sp2, fshape)[fslice].copy()) finally: if not _rfft_mt_safe: _rfft_lock.release() else: # If we're here, it's either because we need a complex result, or we # failed to acquire _rfft_lock (meaning rfftn isn't threadsafe and # is already in use by another thread). In either case, use the # (threadsafe but slower) SciPy complex-FFT routines instead. sp1 = fftpack.fftn(in1, fshape) sp2 = fftpack.fftn(in2, fshape) ret = fftpack.ifftn(sp1 * sp2)[fslice].copy() if not complex_result: ret = ret.real
if mode == "full": return ret elif mode == "same": return _centered(ret, s1) elif mode == "valid": return _centered(ret, s1 - s2 + 1) else: raise ValueError("Acceptable mode flags are 'valid'," " 'same', or 'full'.")
""" See if a list of arrays are all numeric.
Parameters ---------- ndarrays : array or list of arrays arrays to check if numeric. numeric_kinds : string-like The dtypes of the arrays to be checked. If the dtype.kind of the ndarrays are not in this string the function returns False and otherwise returns True. """ if type(arrays) == ndarray: return arrays.dtype.kind in kinds for array_ in arrays: if array_.dtype.kind not in kinds: return False return True
""" Product of a list of numbers. Faster than np.prod for short lists like array shapes. """ product = 1 for x in iterable: product *= x return product
""" See if using `fftconvolve` or `_correlateND` is faster. The boolean value returned depends on the sizes and shapes of the input values.
The big O ratios were found to hold across different machines, which makes sense as it's the ratio that matters (the effective speed of the computer is found in both big O constants). Regardless, this had been tuned on an early 2015 MacBook Pro with 8GB RAM and an Intel i5 processor. """ if mode == 'full': out_shape = [n + k - 1 for n, k in zip(x.shape, h.shape)] big_O_constant = 10963.92823819 if x.ndim == 1 else 8899.1104874 elif mode == 'same': out_shape = x.shape if x.ndim == 1: if h.size <= x.size: big_O_constant = 7183.41306773 else: big_O_constant = 856.78174111 else: big_O_constant = 34519.21021589 elif mode == 'valid': out_shape = [n - k + 1 for n, k in zip(x.shape, h.shape)] big_O_constant = 41954.28006344 if x.ndim == 1 else 66453.24316434 else: raise ValueError("Acceptable mode flags are 'valid'," " 'same', or 'full'.")
# see whether the Fourier transform convolution method or the direct # convolution method is faster (discussed in scikit-image PR #1792) direct_time = (x.size * h.size * _prod(out_shape)) fft_time = sum(n * math.log(n) for n in (x.shape + h.shape + tuple(out_shape))) return big_O_constant * fft_time < direct_time
""" Reverse array `x` in all dimensions and perform the complex conjugate """ reverse = [slice(None, None, -1)] * x.ndim return x[reverse].conj()
""" See if numpy supports convolution of `volume` and `kernel` (i.e. both are 1D ndarrays and of the appropriate shape). Numpy's 'same' mode uses the size of the larger input, while Scipy's uses the size of the first input.
Invalid mode strings will return False and be caught by the calling func. """ if volume.ndim == kernel.ndim == 1: if mode in ('full', 'valid'): return True elif mode == 'same': return volume.size >= kernel.size else: return False
""" Returns the time the statement/function took, in seconds.
Faster, less precise version of IPython's timeit. `stmt` can be a statement written as a string or a callable.
Will do only 1 loop (like IPython's timeit) with no repetitions (unlike IPython) for very slow functions. For fast functions, only does enough loops to take 5 ms, which seems to produce similar results (on Windows at least), and avoids doing an extraneous cycle that isn't measured.
""" timer = timeit.Timer(stmt, setup)
# determine number of calls per rep so total time for 1 rep >= 5 ms x = 0 for p in range(0, 10): number = 10**p x = timer.timeit(number) # seconds if x >= 5e-3 / 10: # 5 ms for final test, 1/10th that for this one break if x > 1: # second # If it's macroscopic, don't bother with repetitions best = x else: number *= 10 r = timer.repeat(repeat, number) best = min(r)
sec = best / number return sec
""" Find the fastest convolution/correlation method.
This primarily exists to be called during the ``method='auto'`` option in `convolve` and `correlate`, but can also be used when performing many convolutions of the same input shapes and dtypes, determining which method to use for all of them, either to avoid the overhead of the 'auto' option or to use accurate real-world measurements.
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 ----- For large n, ``measure=False`` is accurate and can quickly determine the fastest method to perform the convolution. However, this is not as accurate for small n (when any dimension in the input or output is small).
In practice, we found that this function estimates the faster method up to a multiplicative factor of 5 (i.e., the estimated method is *at most* 5 times slower than the fastest method). The estimation values were tuned on an early 2015 MacBook Pro with 8GB RAM but we found that the prediction held *fairly* accurately across different machines.
If ``measure=True``, time the convolutions. Because this function uses `fftconvolve`, an error will be thrown if it does not support the inputs. 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 >>> a = np.random.randn(1000) >>> b = np.random.randn(1000000) >>> method = signal.choose_conv_method(a, b, mode='same') >>> method 'fft'
This can then be applied to other arrays of the same dtype and shape:
>>> c = np.random.randn(1000) >>> d = np.random.randn(1000000) >>> # `method` works with correlate and convolve >>> corr1 = signal.correlate(a, b, mode='same', method=method) >>> corr2 = signal.correlate(c, d, mode='same', method=method) >>> conv1 = signal.convolve(a, b, mode='same', method=method) >>> conv2 = signal.convolve(c, d, mode='same', method=method)
""" volume = asarray(in1) kernel = asarray(in2)
if measure: times = {} for method in ['fft', 'direct']: times[method] = _timeit_fast(lambda: convolve(volume, kernel, mode=mode, method=method))
chosen_method = 'fft' if times['fft'] < times['direct'] else 'direct' return chosen_method, times
# fftconvolve doesn't support complex256 fftconv_unsup = "complex256" if sys.maxsize > 2**32 else "complex192" if hasattr(np, fftconv_unsup): if volume.dtype == fftconv_unsup or kernel.dtype == fftconv_unsup: return 'direct'
# for integer input, # catch when more precision required than float provides (representing an # integer as float can lose precision in fftconvolve if larger than 2**52) if any([_numeric_arrays([x], kinds='ui') for x in [volume, kernel]]): max_value = int(np.abs(volume).max()) * int(np.abs(kernel).max()) max_value *= int(min(volume.size, kernel.size)) if max_value > 2**np.finfo('float').nmant - 1: return 'direct'
if _numeric_arrays([volume, kernel], kinds='b'): return 'direct'
if _numeric_arrays([volume, kernel]): if _fftconv_faster(volume, kernel, mode): return 'fft'
return 'direct'
""" 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
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()
""" volume = asarray(in1) kernel = asarray(in2)
if volume.ndim == kernel.ndim == 0: return volume * kernel elif volume.ndim != kernel.ndim: raise ValueError("volume and kernel should have the same " "dimensionality")
if _inputs_swap_needed(mode, volume.shape, kernel.shape): # Convolution is commutative; order doesn't have any effect on output volume, kernel = kernel, volume
if method == 'auto': method = choose_conv_method(volume, kernel, mode=mode)
if method == 'fft': out = fftconvolve(volume, kernel, mode=mode) result_type = np.result_type(volume, kernel) if result_type.kind in {'u', 'i'}: out = np.around(out) return out.astype(result_type) elif method == 'direct': # fastpath to faster numpy.convolve for 1d inputs when possible if _np_conv_ok(volume, kernel, mode): return np.convolve(volume, kernel, mode)
return correlate(volume, _reverse_and_conj(kernel), mode, 'direct') else: raise ValueError("Acceptable method flags are 'auto'," " 'direct', or 'fft'.")
""" 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.]])
""" domain = asarray(domain) size = domain.shape for k in range(len(size)): if (size[k] % 2) != 1: raise ValueError("Each dimension of domain argument " " should have an odd number of elements.") return sigtools._order_filterND(a, domain, rank)
""" 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`.
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.
""" volume = atleast_1d(volume) if kernel_size is None: kernel_size = [3] * volume.ndim kernel_size = asarray(kernel_size) if kernel_size.shape == (): kernel_size = np.repeat(kernel_size.item(), volume.ndim)
for k in range(volume.ndim): if (kernel_size[k] % 2) != 1: raise ValueError("Each element of kernel_size should be odd.")
domain = ones(kernel_size)
numels = product(kernel_size, axis=0) order = numels // 2 return sigtools._order_filterND(volume, domain, order)
""" 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`.
""" im = asarray(im) if mysize is None: mysize = [3] * im.ndim mysize = asarray(mysize) if mysize.shape == (): mysize = np.repeat(mysize.item(), im.ndim)
# Estimate the local mean lMean = correlate(im, ones(mysize), 'same') / product(mysize, axis=0)
# Estimate the local variance lVar = (correlate(im ** 2, ones(mysize), 'same') / product(mysize, axis=0) - lMean ** 2)
# Estimate the noise power if needed. if noise is None: noise = mean(ravel(lVar), axis=0)
res = (im - lMean) res *= (1 - noise / lVar) res += lMean out = where(lVar < noise, lMean, res)
return out
""" 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()
""" in1 = asarray(in1) in2 = asarray(in2)
if not in1.ndim == in2.ndim == 2: raise ValueError('convolve2d inputs must both be 2D arrays')
if _inputs_swap_needed(mode, in1.shape, in2.shape): in1, in2 = in2, in1
val = _valfrommode(mode) bval = _bvalfromboundary(boundary) out = sigtools._convolve2d(in1, in2, 1, val, bval, fillvalue) return out
""" 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(face[300:365, 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()
""" in1 = asarray(in1) in2 = asarray(in2)
if not in1.ndim == in2.ndim == 2: raise ValueError('correlate2d inputs must both be 2D arrays')
swapped_inputs = _inputs_swap_needed(mode, in1.shape, in2.shape) if swapped_inputs: in1, in2 = in2, in1
val = _valfrommode(mode) bval = _bvalfromboundary(boundary) out = sigtools._convolve2d(in1, in2.conj(), 0, val, bval, fillvalue)
if swapped_inputs: out = out[::-1, ::-1]
return out
""" 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).
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.
""" image = asarray(input) if kernel_size is None: kernel_size = [3] * 2 kernel_size = asarray(kernel_size) if kernel_size.shape == (): kernel_size = np.repeat(kernel_size.item(), 2)
for size in kernel_size: if (size % 2) != 1: raise ValueError("Each element of kernel_size should be odd.")
return sigtools._medfilt2d(image, kernel_size)
""" 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).
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 ``a[0]`` is not 1, then both `a` and `b` are normalized by ``a[0]``. 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::
a[0]*y[n] = b[0]*x[n] + b[1]*x[n-1] + ... + b[M]*x[n-M] - a[1]*y[n-1] - ... - a[N]*y[n-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)::
a[0]*y[n] = b[0] * x[n] + d[0][n-1] d[0][n] = b[1] * x[n] - a[1] * y[n] + d[1][n-1] d[1][n] = b[2] * x[n] - a[2] * y[n] + d[2][n-1] ... d[N-2][n] = b[N-1]*x[n] - a[N-1]*y[n] + d[N-1][n-1] d[N-1][n] = b[N] * x[n] - a[N] * y[n]
where `d` are the state variables.
The rational transfer function describing this filter in the z-transform domain is::
-1 -M b[0] + b[1]z + ... + b[M] z Y(z) = -------------------------------- X(z) -1 -N a[0] + a[1]z + ... + a[N] 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*xn[0])
Apply the filter again, to have a result filtered at an order the same as filtfilt:
>>> z2, _ = signal.lfilter(b, a, z, zi=zi*z[0])
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()
""" # This path only supports types fdgFDGO to mirror _linear_filter below. # Any of b, a, x, or zi can set the dtype, but there is no default # casting of other types; instead a NotImplementedError is raised. raise ValueError('object of too small depth for desired array') # _linear_filter does not broadcast zi, but does do expansion of # singleton dims. raise ValueError('object of too small depth for desired array') # check the trivial case where zi is the right shape first strides = zi.ndim * [None] if axis < 0: axis += zi.ndim for k in range(zi.ndim): if k == axis and zi.shape[k] == expected_shape[k]: strides[k] = zi.strides[k] elif k != axis and zi.shape[k] == expected_shape[k]: strides[k] = zi.strides[k] elif k != axis and zi.shape[k] == 1: strides[k] = 0 else: raise ValueError('Unexpected shape for zi: expected ' '%s, found %s.' % (expected_shape, zi.shape)) zi = np.lib.stride_tricks.as_strided(zi, expected_shape, strides)
raise NotImplementedError("input type '%s' not supported" % dtype)
return out else: else: if zi is None: return sigtools._linear_filter(b, a, x, axis) else: return sigtools._linear_filter(b, a, x, axis, zi)
""" 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
""" N = np.size(a) - 1 M = np.size(b) - 1 K = max(M, N) y = asarray(y) if y.dtype.kind in 'bui': # ensure calculations are floating point y = y.astype(np.float64) zi = zeros(K, y.dtype) if x is None: x = zeros(M, y.dtype) else: x = asarray(x) L = np.size(x) if L < M: x = r_[x, zeros(M - L)] L = np.size(y) if L < N: y = r_[y, zeros(N - L)]
for m in range(M): zi[m] = np.sum(b[m + 1:] * x[:M - m], axis=0)
for m in range(N): zi[m] -= np.sum(a[m + 1:] * y[:N - m], axis=0)
return zi
"""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)
""" num = atleast_1d(signal) den = atleast_1d(divisor) N = len(num) D = len(den) if D > N: quot = [] rem = num else: input = zeros(N - D + 1, float) input[0] = 1 quot = lfilter(num, den, input) rem = num - convolve(den, quot, mode='full') return quot, rem
""" 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.shape[axis]`` 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`
See Also -------- scipy.fftpack.hilbert : Return Hilbert transform of a periodic sequence x.
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(t[1:], instantaneous_frequency) >>> ax1.set_xlabel("time in seconds") >>> ax1.set_ylim(0.0, 120.0)
References ---------- .. [1] Wikipedia, "Analytic signal". http://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
""" x = asarray(x) if iscomplexobj(x): raise ValueError("x must be real.") if N is None: N = x.shape[axis] if N <= 0: raise ValueError("N must be positive.")
Xf = fftpack.fft(x, N, axis=axis) h = zeros(N) if N % 2 == 0: h[0] = h[N // 2] = 1 h[1:N // 2] = 2 else: h[0] = 1 h[1:(N + 1) // 2] = 2
if x.ndim > 1: ind = [newaxis] * x.ndim ind[axis] = slice(None) h = h[ind] x = fftpack.ifft(Xf * h, axis=axis) return x
""" 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", http://en.wikipedia.org/wiki/Analytic_signal
""" x = atleast_2d(x) if x.ndim > 2: raise ValueError("x must be 2-D.") if iscomplexobj(x): raise ValueError("x must be real.") if N is None: N = x.shape elif isinstance(N, int): if N <= 0: raise ValueError("N must be positive.") N = (N, N) elif len(N) != 2 or np.any(np.asarray(N) <= 0): raise ValueError("When given as a tuple, N must hold exactly " "two positive integers")
Xf = fftpack.fft2(x, N, axes=(0, 1)) h1 = zeros(N[0], 'd') h2 = zeros(N[1], 'd') for p in range(2): h = eval("h%d" % (p + 1)) N1 = N[p] if N1 % 2 == 0: h[0] = h[N1 // 2] = 1 h[1:N1 // 2] = 2 else: h[0] = 1 h[1:(N1 + 1) // 2] = 2 exec("h%d = h" % (p + 1), globals(), locals())
h = h1[:, newaxis] * h2[newaxis, :] k = x.ndim while k > 2: h = h[:, newaxis] k -= 1 x = fftpack.ifft2(Xf * h, axes=(0, 1)) return x
"""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])
""" p = asarray(p) if iscomplexobj(p): indx = argsort(abs(p)) else: indx = argsort(p) return take(p, indx, 0), indx
""" 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. Default is 1e-3. rtype : {'max', 'min, 'avg'}, optional How to determine the returned root if multiple roots are within `tol` of each other.
- 'max': pick the maximum of those roots. - 'min': pick the minimum of those roots. - 'avg': take the average of those roots.
Returns ------- pout : ndarray The list of unique roots, sorted from low to high. mult : ndarray The multiplicity of each root.
Notes ----- 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:
>>> uniq[mult > 1] array([ 1.305])
""" if rtype in ['max', 'maximum']: comproot = np.max elif rtype in ['min', 'minimum']: comproot = np.min elif rtype in ['avg', 'mean']: comproot = np.mean else: raise ValueError("`rtype` must be one of " "{'max', 'maximum', 'min', 'minimum', 'avg', 'mean'}") p = asarray(p) * 1.0 tol = abs(tol) p, indx = cmplx_sort(p) pout = [] mult = [] indx = -1 curp = p[0] + 5 * tol sameroots = [] for k in range(len(p)): tr = p[k] if abs(tr - curp) < tol: sameroots.append(tr) curp = comproot(sameroots) pout[indx] = curp mult[indx] += 1 else: pout.append(tr) curp = tr sameroots = [tr] indx += 1 mult.append(1) return array(pout), array(mult)
""" 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) b[0] s**(M) + b[1] s**(M-1) + ... + b[M] H(s) = ------ = ------------------------------------------ a(s) a[0] s**(N) + a[1] s**(N-1) + ... + a[N]
then the partial-fraction expansion H(s) is defined as::
r[0] r[1] r[-1] = -------- + -------- + ... + --------- + k(s) (s-p[0]) (s-p[1]) (s-p[-1])
If there are any repeated roots (closer together than `tol`), then H(s) has terms like::
r[i] r[i+1] r[i+n-1] -------- + ----------- + ... + ----------- (s-p[i]) (s-p[i])**2 (s-p[i])**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. p : array_like Poles. k : array_like Coefficients of the direct polynomial term. tol : float, optional The tolerance for two roots to be considered equal. Default is 1e-3. rtype : {'max', 'min, 'avg'}, optional How to determine the returned root if multiple roots are within `tol` of each other.
- 'max': pick the maximum of those roots. - 'min': pick the minimum of those roots. - 'avg': take the average of those roots.
Returns ------- b : ndarray Numerator polynomial coefficients. a : ndarray Denominator polynomial coefficients.
See Also -------- residue, invresz, unique_roots
""" extra = k p, indx = cmplx_sort(p) r = take(r, indx, 0) pout, mult = unique_roots(p, tol=tol, rtype=rtype) p = [] for k in range(len(pout)): p.extend([pout[k]] * mult[k]) a = atleast_1d(poly(p)) if len(extra) > 0: b = polymul(extra, a) else: b = [0] indx = 0 for k in range(len(pout)): temp = [] for l in range(len(pout)): if l != k: temp.extend([pout[l]] * mult[l]) for m in range(mult[k]): t2 = temp[:] t2.extend([pout[k]] * (mult[k] - m - 1)) b = polyadd(b, r[indx] * atleast_1d(poly(t2))) indx += 1 b = real_if_close(b) while allclose(b[0], 0, rtol=1e-14) and (b.shape[-1] > 1): b = b[1:] return b, a
""" 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) b[0] s**(M) + b[1] s**(M-1) + ... + b[M] H(s) = ------ = ------------------------------------------ a(s) a[0] s**(N) + a[1] s**(N-1) + ... + a[N]
then the partial-fraction expansion H(s) is defined as::
r[0] r[1] r[-1] = -------- + -------- + ... + --------- + k(s) (s-p[0]) (s-p[1]) (s-p[-1])
If there are any repeated roots (closer together than `tol`), then H(s) has terms like::
r[i] r[i+1] r[i+n-1] -------- + ----------- + ... + ----------- (s-p[i]) (s-p[i])**2 (s-p[i])**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`.
Parameters ---------- b : array_like Numerator polynomial coefficients. a : array_like Denominator polynomial coefficients.
Returns ------- r : ndarray Residues. p : ndarray Poles. k : ndarray Coefficients of the direct polynomial term.
See Also -------- invres, residuez, numpy.poly, unique_roots
"""
b, a = map(asarray, (b, a)) rscale = a[0] k, b = polydiv(b, a) p = roots(a) r = p * 0.0 pout, mult = unique_roots(p, tol=tol, rtype=rtype) p = [] for n in range(len(pout)): p.extend([pout[n]] * mult[n]) p = asarray(p) # Compute the residue from the general formula indx = 0 for n in range(len(pout)): bn = b.copy() pn = [] for l in range(len(pout)): if l != n: pn.extend([pout[l]] * mult[l]) an = atleast_1d(poly(pn)) # bn(s) / an(s) is (s-po[n])**Nn * b(s) / a(s) where Nn is # multiplicity of pole at po[n] sig = mult[n] for m in range(sig, 0, -1): if sig > m: # compute next derivative of bn(s) / an(s) term1 = polymul(polyder(bn, 1), an) term2 = polymul(bn, polyder(an, 1)) bn = polysub(term1, term2) an = polymul(an, an) r[indx + m - 1] = (polyval(bn, pout[n]) / polyval(an, pout[n]) / factorial(sig - m)) indx += sig return r / rscale, p, k
""" 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) b[0] + b[1] z**(-1) + ... + b[M] z**(-M) H(z) = ------ = ------------------------------------------ a(z) a[0] + a[1] z**(-1) + ... + a[N] z**(-N)
then the partial-fraction expansion H(z) is defined as::
r[0] r[-1] = --------------- + ... + ---------------- + k[0] + k[1]z**(-1) ... (1-p[0]z**(-1)) (1-p[-1]z**(-1))
If there are any repeated roots (closer than `tol`), then the partial fraction expansion has terms like::
r[i] r[i+1] r[i+n-1] -------------- + ------------------ + ... + ------------------ (1-p[i]z**(-1)) (1-p[i]z**(-1))**2 (1-p[i]z**(-1))**n
This function is used for polynomials in negative powers of z, such as digital filters in DSP. For positive powers, use `residue`.
Parameters ---------- b : array_like Numerator polynomial coefficients. a : array_like Denominator polynomial coefficients.
Returns ------- r : ndarray Residues. p : ndarray Poles. k : ndarray Coefficients of the direct polynomial term.
See Also -------- invresz, residue, unique_roots
""" b, a = map(asarray, (b, a)) gain = a[0] brev, arev = b[::-1], a[::-1] krev, brev = polydiv(brev, arev) if krev == []: k = [] else: k = krev[::-1] b = brev[::-1] p = roots(a) r = p * 0.0 pout, mult = unique_roots(p, tol=tol, rtype=rtype) p = [] for n in range(len(pout)): p.extend([pout[n]] * mult[n]) p = asarray(p) # Compute the residue from the general formula (for discrete-time) # the polynomial is in z**(-1) and the multiplication is by terms # like this (1-p[i] z**(-1))**mult[i]. After differentiation, # we must divide by (-p[i])**(m-k) as well as (m-k)! indx = 0 for n in range(len(pout)): bn = brev.copy() pn = [] for l in range(len(pout)): if l != n: pn.extend([pout[l]] * mult[l]) an = atleast_1d(poly(pn))[::-1] # bn(z) / an(z) is (1-po[n] z**(-1))**Nn * b(z) / a(z) where Nn is # multiplicity of pole at po[n] and b(z) and a(z) are polynomials. sig = mult[n] for m in range(sig, 0, -1): if sig > m: # compute next derivative of bn(s) / an(s) term1 = polymul(polyder(bn, 1), an) term2 = polymul(bn, polyder(an, 1)) bn = polysub(term1, term2) an = polymul(an, an) r[indx + m - 1] = (polyval(bn, 1.0 / pout[n]) / polyval(an, 1.0 / pout[n]) / factorial(sig - m) / (-pout[n]) ** (sig - m)) indx += sig return r / gain, p, k
""" 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) b[0] + b[1] z**(-1) + ... + b[M] z**(-M) H(z) = ------ = ------------------------------------------ a(z) a[0] + a[1] z**(-1) + ... + a[N] z**(-N)
then the partial-fraction expansion H(z) is defined as::
r[0] r[-1] = --------------- + ... + ---------------- + k[0] + k[1]z**(-1) ... (1-p[0]z**(-1)) (1-p[-1]z**(-1))
If there are any repeated roots (closer than `tol`), then the partial fraction expansion has terms like::
r[i] r[i+1] r[i+n-1] -------------- + ------------------ + ... + ------------------ (1-p[i]z**(-1)) (1-p[i]z**(-1))**2 (1-p[i]z**(-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. p : array_like Poles. k : array_like Coefficients of the direct polynomial term. tol : float, optional The tolerance for two roots to be considered equal. Default is 1e-3. rtype : {'max', 'min, 'avg'}, optional How to determine the returned root if multiple roots are within `tol` of each other.
- 'max': pick the maximum of those roots. - 'min': pick the minimum of those roots. - 'avg': take the average of those roots.
Returns ------- b : ndarray Numerator polynomial coefficients. a : ndarray Denominator polynomial coefficients.
See Also -------- residuez, unique_roots, invres
""" extra = asarray(k) p, indx = cmplx_sort(p) r = take(r, indx, 0) pout, mult = unique_roots(p, tol=tol, rtype=rtype) p = [] for k in range(len(pout)): p.extend([pout[k]] * mult[k]) a = atleast_1d(poly(p)) if len(extra) > 0: b = polymul(extra, a) else: b = [0] indx = 0 brev = asarray(b)[::-1] for k in range(len(pout)): temp = [] # Construct polynomial which does not include any of this root for l in range(len(pout)): if l != k: temp.extend([pout[l]] * mult[l]) for m in range(mult[k]): t2 = temp[:] t2.extend([pout[k]] * (mult[k] - m - 1)) brev = polyadd(brev, (r[indx] * atleast_1d(poly(t2)))[::-1]) indx += 1 b = real_if_close(brev[::-1]) return b, a
""" 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 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.shape[axis]) ).
If `window` is an array of the same length as `x.shape[axis]` 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 represents the old sample positions, and the new sample positions will be returned as well as the new samples.
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.fftpack.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, y[0], 'ro') >>> plt.legend(['data', 'resampled'], loc='best') >>> plt.show() """ x = asarray(x) X = fftpack.fft(x, axis=axis) Nx = x.shape[axis] if window is not None: if callable(window): W = window(fftpack.fftfreq(Nx)) elif isinstance(window, ndarray): if window.shape != (Nx,): raise ValueError('window must have the same length as data') W = window else: W = fftpack.ifftshift(get_window(window, Nx)) newshape = [1] * x.ndim newshape[axis] = len(W) W.shape = newshape X = X * W W.shape = (Nx,) sl = [slice(None)] * x.ndim newshape = list(x.shape) newshape[axis] = num N = int(np.minimum(num, Nx)) Y = zeros(newshape, 'D') sl[axis] = slice(0, (N + 1) // 2) Y[sl] = X[sl] sl[axis] = slice(-(N - 1) // 2, None) Y[sl] = X[sl]
if N % 2 == 0: # special treatment if low number of points is even. So far we have set Y[-N/2]=X[-N/2] if N < Nx: # if downsampling sl[axis] = slice(N//2,N//2+1,None) # select the component at frequency N/2 Y[sl] += X[sl] # add the component of X at N/2 elif N < num: # if upsampling sl[axis] = slice(num-N//2,num-N//2+1,None) # select the component at frequency -N/2 Y[sl] /= 2 # halve the component at -N/2 temp = Y[sl] sl[axis] = slice(N//2,N//2+1,None) # select the component at +N/2 Y[sl] = temp # set that equal to the component at -N/2
y = fftpack.ifft(Y, axis=axis) * (float(num) / float(Nx))
if x.dtype.char not in ['F', 'D']: y = y.real
if t is None: return y else: new_t = arange(0, num) * (t[1] - t[0]) * Nx / float(num) + t[0] return y, new_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. 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.
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 -------- Note that 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, y[0], 'bo', 10, 0., 'ro') # boundaries >>> plt.legend(['resample', 'resamp_poly', 'data'], loc='best') >>> plt.show() """ x = asarray(x) if up != int(up): raise ValueError("up must be an integer") if down != int(down): raise ValueError("down must be an integer") up = int(up) down = int(down) if up < 1 or down < 1: raise ValueError('up and down must be >= 1')
# Determine our up and down factors # Use a rational approimation to save computation time on really long # signals g_ = gcd(up, down) up //= g_ down //= g_ if up == down == 1: return x.copy() n_out = x.shape[axis] * up n_out = n_out // down + bool(n_out % down)
if isinstance(window, (list, np.ndarray)): window = array(window) # use array to force a copy (we modify it) if window.ndim > 1: raise ValueError('window must be 1-D') half_len = (window.size - 1) // 2 h = window else: # Design a linear-phase low-pass FIR filter max_rate = max(up, down) f_c = 1. / max_rate # cutoff of FIR filter (rel. to Nyquist) half_len = 10 * max_rate # reasonable cutoff for our sinc-like function h = firwin(2 * half_len + 1, f_c, window=window) h *= up
# Zero-pad our filter to put the output samples at the center n_pre_pad = (down - half_len % down) n_post_pad = 0 n_pre_remove = (half_len + n_pre_pad) // down # We should rarely need to do this given our filter lengths... while _output_len(len(h) + n_pre_pad + n_post_pad, x.shape[axis], up, down) < n_out + n_pre_remove: n_post_pad += 1 h = np.concatenate((np.zeros(n_pre_pad), h, np.zeros(n_post_pad))) n_pre_remove_end = n_pre_remove + n_out
# filter then remove excess y = upfirdn(h, x, up, down, axis=axis) keep = [slice(None), ]*x.ndim keep[axis] = slice(n_pre_remove, n_pre_remove_end) return y[keep]
''' 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`. ''' events = asarray(events) period = asarray(period) if events.ndim > 1: raise ValueError('events cannot have dimensions more than 1') if period.ndim > 1: raise ValueError('period cannot have dimensions more than 1')
# we need to know later if period was originally a scalar scalarperiod = not period.ndim
events = atleast_2d(events) period = atleast_2d(period) if (period <= 0).any(): raise ValueError('periods must be positive')
# this converts the times to vectors vectors = exp(dot(2j*pi/period.T, events))
# the vector strength is just the magnitude of the mean of the vectors # the vector phase is the angle of the mean of the vectors vectormean = mean(vectors, axis=1) strength = abs(vectormean) phase = angle(vectormean)
# if the original period was a scalar, return scalars if scalarperiod: strength = strength[0] phase = phase[0] return strength, phase
""" 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`.
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
""" if type not in ['linear', 'l', 'constant', 'c']: raise ValueError("Trend type must be 'linear' or 'constant'.") data = asarray(data) dtype = data.dtype.char if dtype not in 'dfDF': dtype = 'd' if type in ['constant', 'c']: ret = data - expand_dims(mean(data, axis), axis) return ret else: dshape = data.shape N = dshape[axis] bp = sort(unique(r_[0, bp, N])) if np.any(bp > N): raise ValueError("Breakpoints must be less than length " "of data along given axis.") Nreg = len(bp) - 1 # Restructure data so that axis is along first dimension and # all other dimensions are collapsed into second dimension rnk = len(dshape) if axis < 0: axis = axis + rnk newdims = r_[axis, 0:axis, axis + 1:rnk] newdata = reshape(transpose(data, tuple(newdims)), (N, _prod(dshape) // N)) newdata = newdata.copy() # make sure we have a copy if newdata.dtype.char not in 'dfDF': newdata = newdata.astype(dtype) # Find leastsq fit and remove it for each piece for m in range(Nreg): Npts = bp[m + 1] - bp[m] A = ones((Npts, 2), dtype) A[:, 0] = cast[dtype](arange(1, Npts + 1) * 1.0 / Npts) sl = slice(bp[m], bp[m + 1]) coef, resids, rank, s = linalg.lstsq(A, newdata[sl]) newdata[sl] = newdata[sl] - dot(A, coef) # Put data back in original shape. tdshape = take(dshape, newdims, 0) ret = reshape(newdata, tuple(tdshape)) vals = list(range(1, rnk)) olddims = vals[:axis] + [0] + vals[axis:] ret = transpose(ret, tuple(olddims)) return ret
""" 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 = b[1:] - a[1:]*b[0]
assuming `a[0]` is 1.0; if `a[0]` is not 1, `a` and `b` are first divided by a[0].
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*x[0]) >>> 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 `x[0]`. Then the output `y` has no transient until the input drops from 0.5 to 0.0.
"""
# FIXME: Can this function be replaced with an appropriate # use of lfiltic? For example, when b,a = butter(N,Wn), # lfiltic(b, a, y=numpy.ones_like(a), x=numpy.ones_like(b)). #
# We could use scipy.signal.normalize, but it uses warnings in # cases where a ValueError is more appropriate, and it allows # b to be 2D. raise ValueError("Numerator b must be 1-D.") raise ValueError("Denominator a must be 1-D.")
a = a[1:] raise ValueError("There must be at least one nonzero `a` coefficient.")
# Normalize the coefficients so a[0] == 1. b = b / a[0] a = a / a[0]
# Pad a or b with zeros so they are the same length. elif len(b) < n: b = np.r_[b, np.zeros(n - len(b))]
# Solve zi = A*zi + B
# For future reference: we could also use the following # explicit formulas to solve the linear system: # # zi = np.zeros(n - 1) # zi[0] = B.sum() / IminusA[:,0].sum() # asum = 1.0 # csum = 0.0 # for k in range(1,n-1): # asum += a[k] # csum += b[k] - a[k]*b[0] # zi[k] = asum*zi[0] - csum
""" 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()
""" sos = np.asarray(sos) if sos.ndim != 2 or sos.shape[1] != 6: raise ValueError('sos must be shape (n_sections, 6)')
n_sections = sos.shape[0] zi = np.empty((n_sections, 2)) scale = 1.0 for section in range(n_sections): b = sos[section, :3] a = sos[section, 3:] zi[section] = scale * lfilter_zi(b, a) # If H(z) = B(z)/A(z) is this section's transfer function, then # b.sum()/a.sum() is H(1), the gain at omega=0. That's the steady # state value of this section's step response. scale *= b.sum() / a.sum()
return zi
"""Forward-backward IIR filter that uses Gustafsson's method.
Apply the IIR filter defined by `(b,a)` to `x` twice, first forward then backward, using Gustafsson's initial conditions [1]_.
Let ``y_fb`` be the result of filtering first forward and then backward, and let ``y_bf`` be the result of filtering first backward then forward. Gustafsson's method is to compute initial conditions for the forward pass and the backward pass such that ``y_fb == y_bf``.
Parameters ---------- b : scalar or 1-D ndarray Numerator coefficients of the filter. a : scalar or 1-D ndarray Denominator coefficients of the filter. x : ndarray Data to be filtered. axis : int, optional Axis of `x` to be filtered. Default is -1. irlen : int or None, optional The length of the nonnegligible part of the impulse response. If `irlen` is None, or if the length of the signal is less than ``2 * irlen``, then no part of the impulse response is ignored.
Returns ------- y : ndarray The filtered data. x0 : ndarray Initial condition for the forward filter. x1 : ndarray Initial condition for the backward filter.
Notes ----- Typically the return values `x0` and `x1` are not needed by the caller. The intended use of these return values is in unit tests.
References ---------- .. [1] F. Gustaffson. Determining the initial states in forward-backward filtering. Transactions on Signal Processing, 46(4):988-992, 1996.
""" # In the comments, "Gustafsson's paper" and [1] refer to the # paper referenced in the docstring.
b = np.atleast_1d(b) a = np.atleast_1d(a)
order = max(len(b), len(a)) - 1 if order == 0: # The filter is just scalar multiplication, with no state. scale = (b[0] / a[0])**2 y = scale * x return y, np.array([]), np.array([])
if axis != -1 or axis != x.ndim - 1: # Move the axis containing the data to the end. x = np.swapaxes(x, axis, x.ndim - 1)
# n is the number of samples in the data to be filtered. n = x.shape[-1]
if irlen is None or n <= 2*irlen: m = n else: m = irlen
# Create Obs, the observability matrix (called O in the paper). # This matrix can be interpreted as the operator that propagates # an arbitrary initial state to the output, assuming the input is # zero. # In Gustafsson's paper, the forward and backward filters are not # necessarily the same, so he has both O_f and O_b. We use the same # filter in both directions, so we only need O. The same comment # applies to S below. Obs = np.zeros((m, order)) zi = np.zeros(order) zi[0] = 1 Obs[:, 0] = lfilter(b, a, np.zeros(m), zi=zi)[0] for k in range(1, order): Obs[k:, k] = Obs[:-k, 0]
# Obsr is O^R (Gustafsson's notation for row-reversed O) Obsr = Obs[::-1]
# Create S. S is the matrix that applies the filter to the reversed # propagated initial conditions. That is, # out = S.dot(zi) # is the same as # tmp, _ = lfilter(b, a, zeros(), zi=zi) # Propagate ICs. # out = lfilter(b, a, tmp[::-1]) # Reverse and filter.
# Equations (5) & (6) of [1] S = lfilter(b, a, Obs[::-1], axis=0)
# Sr is S^R (row-reversed S) Sr = S[::-1]
# M is [(S^R - O), (O^R - S)] if m == n: M = np.hstack((Sr - Obs, Obsr - S)) else: # Matrix described in section IV of [1]. M = np.zeros((2*m, 2*order)) M[:m, :order] = Sr - Obs M[m:, order:] = Obsr - S
# Naive forward-backward and backward-forward filters. # These have large transients because the filters use zero initial # conditions. y_f = lfilter(b, a, x) y_fb = lfilter(b, a, y_f[..., ::-1])[..., ::-1]
y_b = lfilter(b, a, x[..., ::-1])[..., ::-1] y_bf = lfilter(b, a, y_b)
delta_y_bf_fb = y_bf - y_fb if m == n: delta = delta_y_bf_fb else: start_m = delta_y_bf_fb[..., :m] end_m = delta_y_bf_fb[..., -m:] delta = np.concatenate((start_m, end_m), axis=-1)
# ic_opt holds the "optimal" initial conditions. # The following code computes the result shown in the formula # of the paper between equations (6) and (7). if delta.ndim == 1: ic_opt = linalg.lstsq(M, delta)[0] else: # Reshape delta so it can be used as an array of multiple # right-hand-sides in linalg.lstsq. delta2d = delta.reshape(-1, delta.shape[-1]).T ic_opt0 = linalg.lstsq(M, delta2d)[0].T ic_opt = ic_opt0.reshape(delta.shape[:-1] + (M.shape[-1],))
# Now compute the filtered signal using equation (7) of [1]. # First, form [S^R, O^R] and call it W. if m == n: W = np.hstack((Sr, Obsr)) else: W = np.zeros((2*m, 2*order)) W[:m, :order] = Sr W[m:, order:] = Obsr
# Equation (7) of [1] says # Y_fb^opt = Y_fb^0 + W * [x_0^opt; x_{N-1}^opt] # `wic` is (almost) the product on the right. # W has shape (m, 2*order), and ic_opt has shape (..., 2*order), # so we can't use W.dot(ic_opt). Instead, we dot ic_opt with W.T, # so wic has shape (..., m). wic = ic_opt.dot(W.T)
# `wic` is "almost" the product of W and the optimal ICs in equation # (7)--if we're using a truncated impulse response (m < n), `wic` # contains only the adjustments required for the ends of the signal. # Here we form y_opt, taking this into account if necessary. y_opt = y_fb if m == n: y_opt += wic else: y_opt[..., :m] += wic[..., :m] y_opt[..., -m:] += wic[..., -m:]
x0 = ic_opt[..., :order] x1 = ic_opt[..., -order:] if axis != -1 or axis != x.ndim - 1: # Restore the data axis to its original position. x0 = np.swapaxes(x0, axis, x.ndim - 1) x1 = np.swapaxes(x1, axis, x.ndim - 1) y_opt = np.swapaxes(y_opt, axis, x.ndim - 1)
return y_opt, x0, x1
irlen=None): """ 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.
Parameters ---------- b : (N,) array_like The numerator coefficient vector of the filter. a : (N,) array_like The denominator coefficient vector of the filter. If ``a[0]`` is not 1, then both `a` and `b` are normalized by ``a[0]``. 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.shape[axis] - 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
"""
raise ValueError("method must be 'pad' or 'gust'.")
y, z1, z2 = _filtfilt_gust(b, a, x, axis=axis, irlen=irlen) return y
# method == "pad" ntaps=max(len(a), len(b)))
# Get the steady state of the filter's step response.
# Reshape zi and create x0 so that zi*x0 broadcasts # to the correct value for the 'zi' keyword argument # to lfilter.
# Forward filter.
# Backward filter. # Create y0 so zi*y0 broadcasts appropriately.
# Reverse y.
# Slice the actual signal from the extended signal.
"""Helper to validate padding for filtfilt""" raise ValueError(("Unknown value '%s' given to padtype. padtype " "must be 'even', 'odd', 'constant', or None.") % padtype)
padlen = 0
# Original padding; preserved for backwards compatibility. else: edge = padlen
# x's 'axis' dimension must be bigger than edge. raise ValueError("The length of the input vector x must be at least " "padlen, which is %d." % edge)
# Make an extension of length `edge` at each # end of the input array. ext = even_ext(x, edge, axis=axis) else: ext = const_ext(x, edge, axis=axis) else: ext = x
""" Filter data along one dimension using cascaded second-order sections.
Filter a data sequence, `x`, using a digital IIR filter defined by `sos`. This is implemented by performing `lfilter` for each second-order section. See `lfilter` for details.
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.shape[axis]`` 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()
""" x = np.asarray(x) sos, n_sections = _validate_sos(sos) use_zi = zi is not None if use_zi: zi = np.asarray(zi) x_zi_shape = list(x.shape) x_zi_shape[axis] = 2 x_zi_shape = tuple([n_sections] + x_zi_shape) if zi.shape != x_zi_shape: raise ValueError('Invalid zi shape. With axis=%r, an input with ' 'shape %r, and an sos array with %d sections, zi ' 'must have shape %r, got %r.' % (axis, x.shape, n_sections, x_zi_shape, zi.shape)) zf = zeros_like(zi)
for section in range(n_sections): if use_zi: x, zf[section] = lfilter(sos[section, :3], sos[section, 3:], x, axis, zi=zi[section]) else: x = lfilter(sos[section, :3], sos[section, 3:], x, axis) out = (x, zf) if use_zi else x return out
""" 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.shape[axis] - 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()
""" sos, n_sections = _validate_sos(sos)
# `method` is "pad"... ntaps = 2 * n_sections + 1 ntaps -= min((sos[:, 2] == 0).sum(), (sos[:, 5] == 0).sum()) edge, ext = _validate_pad(padtype, padlen, x, axis, ntaps=ntaps)
# These steps follow the same form as filtfilt with modifications zi = sosfilt_zi(sos) # shape (n_sections, 2) --> (n_sections, ..., 2, ...) zi_shape = [1] * x.ndim zi_shape[axis] = 2 zi.shape = [n_sections] + zi_shape x_0 = axis_slice(ext, stop=1, axis=axis) (y, zf) = sosfilt(sos, ext, axis=axis, zi=zi * x_0) y_0 = axis_slice(y, start=-1, axis=axis) (y, zf) = sosfilt(sos, axis_reverse(y, axis=axis), axis=axis, zi=zi * y_0) y = axis_reverse(y, axis=axis) if edge > 0: y = axis_slice(y, start=edge, stop=-edge, axis=axis) return y
""" 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. """
x = asarray(x) q = operator.index(q)
if n is not None: n = operator.index(n)
if ftype == 'fir': if n is None: half_len = 10 * q # reasonable cutoff for our sinc-like function n = 2 * half_len b, a = firwin(n+1, 1. / q, window='hamming'), 1. elif ftype == 'iir': if n is None: n = 8 system = dlti(*cheby1(n, 0.05, 0.8 / q)) b, a = system.num, system.den elif isinstance(ftype, dlti): system = ftype._as_tf() # Avoids copying if already in TF form b, a = system.num, system.den else: raise ValueError('invalid ftype')
sl = [slice(None)] * x.ndim a = np.asarray(a)
if a.size == 1: # FIR case b = b / a if zero_phase: y = resample_poly(x, 1, q, axis=axis, window=b) else: # upfirdn is generally faster than lfilter by a factor equal to the # downsampling factor, since it only calculates the needed outputs n_out = x.shape[axis] // q + bool(x.shape[axis] % q) y = upfirdn(b, x, up=1, down=q, axis=axis) sl[axis] = slice(None, n_out, None)
else: # IIR case if zero_phase: y = filtfilt(b, a, x, axis=axis) else: y = lfilter(b, a, x, axis=axis) sl[axis] = slice(None, None, q)
return y[sl] |