import numpy as np
from scipy.special import iv as bessel_first_mod
from scipy.interpolate import interp1d
import scipy.signal as sgn
import matplotlib.pyplot as plt
import pyfar as pf
from scipy.ndimage import generic_filter1d
from fractions import Fraction
from decimal import Decimal
import warnings
def _weighted_moving_average(input, output, weights):
"""Moving average filter of length N and arbitrary.
Parameters
----------
input : numpy.ndarray
The input array
output : numpy.ndarray
The output buffer
N : int
Length of the filter
weights : numpy.ndarray
The weights used for averaging. The length of the weights also
specifies the length of the filter.
Note
----
This function is primarily intended to be used in combination with
``scipy.ndimage.generic_filter1d``. The input is strided instead of
reshaped, leaving the memory layout unchanged. The function does also not
return it's output but requires the output buffer as function input, which
is required by ``scipy.ndimage.generic_filter1d``.
"""
strided = np.lib.stride_tricks.as_strided(
input, strides=input.strides*2,
shape=(weights.size, input.size - (weights.size-1)))
output[:] = np.average(strided, weights=weights, axis=0)
[docs]
def smooth_fractional_octave(signal, num_fractions, mode="magnitude_zerophase",
window="boxcar"):
"""
Smooth spectrum with a fractional octave width.
The smoothing is done according to Tylka et al. 2017 [#]_ (method 2) in
three steps:
1. Interpolate the spectrum to a logarithmically spaced frequency scale
2. Smooth the spectrum by convolution with a smoothing window
3. Interpolate the spectrum to the original linear frequency scale
Parameters
----------
signal : pyfar.Signal
The input data.
num_fractions : number
The width of the smoothing window in fractional octaves, e.g., 3 will
apply third octave smoothing and 1 will apply octave smoothing.
mode : str, optional
``"magnitude_zerophase"``
Only the magnitude response, i.e., the absolute spectrum is
smoothed. Note that this return a zero-phase signal. It might be
necessary to generate a minimum or linear phase if the data is
subject to further processing after the smoothing (cf.
:py:func:`~pyfar.dsp.minimum_phase` and
:py:func:`~pyfar.dsp.linear_phase`)
``"magnitude"``
Smooth the magnitude and keep the phase of the input signal.
``"magnitude_phase"``
Separately smooth the magnitude and unwrapped phase response.
``"complex"``
Separately smooth the real and imaginary part of the spectrum.
Note that the modes `magnitude_zerophase` and `magnitude` make sure
that the smoothed magnitude response is as expected at the cost of an
artificial phase response. This is often desired, e.g., when plotting
signals or designing compensation filters. The modes `magnitude_phase`
and `complex` smooth all information but might cause a high frequency
energy loss in the smoothed magnitude response. The default is
``"magnitude_zerophase"``.
window : str, optional
String that defines the smoothing window. All windows from
:py:func:`~pyfar.dsp.time_window` that do not require an additional
parameter can be used. The default is "boxcar", which uses the
most commonly used rectangular window.
Returns
-------
signal : pyfar.Signal
The smoothed output data
window_stats : tuple
A tuple containing information about the smoothing process
`n_window`
The window length in (logarithmically spaced) samples
`num_fractions`
The actual width of the window in fractional octaves. This can
deviate from the desired width because the smoothing window must
have an integer sample length
Notes
-----
Method 3 in Tylka at al. 2017 is mathematically more elegant at the
price of a largely increased computational and memory cost. In most
practical cases, methods 2 and 3 yield close to identical results (cf. Fig.
2 and 3 in Tylka et al. 2017). If the spectrum contains extreme
discontinuities, however, method 3 is superior (see examples below).
References
----------
.. [#] J. G. Tylka, B. B. Boren, and E. Y. Choueiri, “A Generalized Method
for Fractional-Octave Smoothing of Transfer Functions that Preserves
Log-Frequency Symmetry (Engineering Report),” J. Audio Eng. Soc. 65,
239-245 (2017). doi:10.17743/jaes.2016.0053
Examples
--------
Octave smoothing of continuous spectrum consisting of two bell filters.
.. plot::
>>> import pyfar as pf
>>> signal = pf.signals.impulse(441)
>>> signal = pf.dsp.filter.bell(signal, 1e3, 12, 1, "III")
>>> signal = pf.dsp.filter.bell(signal, 10e3, -60, 100, "III")
>>> smoothed, _ = pf.dsp.smooth_fractional_octave(signal, 1)
>>> ax = pf.plot.freq(signal, label="input")
>>> pf.plot.freq(smoothed, label="smoothed")
>>> ax.legend(loc=3)
Octave smoothing of the discontinuous spectrum of a sine signal causes
artifacts at the edges due to the intermediate interpolation steps (cf.
Tylka et al. 2017, Fig. 4). However this is a rather unusual application
and is mentioned only for the sake of completeness.
.. plot::
>>> import pyfar as pf
>>> signal = pf.signals.sine(1e3, 4410)
>>> signal.fft_norm = "amplitude"
>>> smoothed, _ = pf.dsp.smooth_fractional_octave(signal, 1)
>>> ax = pf.plot.freq(signal, label="input")
>>> pf.plot.freq(smoothed, label="smoothed")
>>> ax.set_xlim(200, 4e3)
>>> ax.set_ylim(-45, 5)
>>> ax.legend(loc=3)
"""
if not isinstance(signal, pf.Signal):
raise TypeError("Input signal has to be of type pyfar.Signal")
if mode in ["magnitude_zerophase", "magnitude"]:
data = [np.atleast_2d(np.abs(signal.freq_raw))]
elif mode == "complex":
data = [np.atleast_2d(np.real(signal.freq_raw)),
np.atleast_2d(np.imag(signal.freq_raw))]
elif mode == "magnitude_phase":
data = [np.atleast_2d(np.abs(signal.freq_raw)),
np.atleast_2d(pf.dsp.phase(signal, unwrap=True))]
else:
raise ValueError((f"mode is '{mode}' but must be 'magnitude_zerophase'"
", 'magnitude_phase', 'magnitude', or 'complex'"))
# linearly and logarithmically spaced frequency bins ----------------------
N = signal.n_bins
n_lin = np.arange(N)
n_log = N**(n_lin/(N-1))
# frequency bin spacing in octaves: log2(n_log[n]/n_log[n-1])
# Note: n_log[0] = 1
delta_n = np.log2(n_log[1])
# width of the window in logarithmically spaced samples
# Note: Forcing the window to have an odd length increases the deviation
# from the exact width, but makes sure that the delay introduced in
# the convolution is integer and can be easily compensated
n_window = int(2 * np.floor(1 / (num_fractions * delta_n * 2)) + 1)
if n_window == 1:
raise ValueError((
"The smoothing width given by num_fractions is below the frequency"
" resolution of the signal. Increase the signal length or decrease"
" num_fractions"))
# generate the smoothing window
if isinstance(window, str):
window = sgn.windows.get_window(window, n_window, fftbins=False)
elif isinstance(window, (list, np.ndarray)):
# undocumented possibility for testing
window = np.asanyarray(window, dtype=float)
if window.shape != (n_window, ):
raise ValueError(
f"window.shape is {window.shape} but must be ({n_window}, )")
else:
raise ValueError(f"window is of type {str(type(window))} but must be "
"of type string")
for nn in range(len(data)):
# interpolate to logarithmically spaced frequencies
interpolator = interp1d(
n_lin + 1, data[nn], "cubic", copy=False, assume_sorted=True)
data[nn] = interpolator(n_log)
# apply a moving average filter based on the window function
data[nn] = generic_filter1d(
data[nn],
function=_weighted_moving_average,
filter_size=n_window,
mode='nearest',
extra_arguments=(window,))
# interpolate to original frequency axis
interpolator = interp1d(
n_log, data[nn], "cubic", copy=False, assume_sorted=True)
data[nn] = interpolator(n_lin + 1)
# generate return signal --------------------------------------------------
if mode == "magnitude_zerophase":
data = data[0]
elif mode == "complex":
data = data[0] + 1j * data[1]
elif mode == "magnitude_phase":
data = data[0] * np.exp(1j * data[1])
elif mode == "magnitude":
data = data[0] * np.exp(1j * np.angle(signal.freq_raw))
# force 0 Hz and Nyquist to be real if it might not be the case
if mode in ["complex", "magnitude_phase", "magnitude"]:
data[..., 0] = np.abs(data[..., 0])
data[..., -1] = np.abs(data[..., -1])
signal = signal.copy()
signal.freq_raw = data
return signal, (n_window, 1 / (n_window * delta_n))
[docs]
def fractional_time_shift(signal, shift, unit="samples", order=30,
side_lobe_suppression=60, mode="linear"):
"""
Apply fractional time shift to input data.
This function uses a windowed Sinc filter (Method FIR-2 in [#]_ according
to Equations 21 and 22) to apply fractional delays, i.e., non-integer
delays to an input signal. A Kaiser window according to [#]_ Equations
(10.12) and (10.13) is used, which offers the possibility to control the
side lobe suppression.
Parameters
----------
signal : Signal
The input data
shift : float, array like
The fractional shift in samples (positive or negative). If this is a
float, the same shift is applied to all channels of `signal`. If this
is an array like different delays are applied to the channels of
`signal`. In this case it must broadcast to `signal.cshape` (see
`Numpy broadcasting
<https://numpy.org/doc/stable/user/basics.broadcasting.html>`_)
unit : str, optional
The unit of the shift. Either 'samples' or 's'. Defaults to 'samples'.
order : int, optional
The order of the fractional shift (sinc) filter. The precision of the
filter increases with the order. High frequency errors decrease with
increasing order. The order must be smaller than
``signal.n_samples``. The default is ``30``.
side_lobe_suppression : float, optional
The side lobe suppression of the Kaiser window in dB. The default is
``60``.
mode : str, optional
The filtering mode
``"linear"``
Apply linear shift, i.e., parts of the signal that are shifted to
times smaller than 0 samples and larger than ``signal.n_samples``
disappear.
``"cyclic"``
Apply a cyclic shift, i.e., parts of the signal that are shifted to
values smaller than 0 are wrapped around to the end, and parts that
are shifted to values larger than ``signal.n_samples`` are wrapped
around to the beginning.
The default is ``"linear"``
Returns
-------
signal : Signal
The delayed input data
References
----------
.. [#] T. I. Laakso, V. Välimäki, M. Karjalainen, and U. K. Laine,
'Splitting the unit delay,' IEEE Signal Processing Magazine 13,
30-60 (1996). doi:10.1109/79.482137
.. [#] A. V. Oppenheim and R. W. Schafer, Discrete-time signal processing,
(Upper Saddle et al., Pearson, 2010), Third edition.
Examples
--------
Apply a fractional shift of 2.3 samples using filters of orders 6 and 30
.. plot::
>>> import pyfar as pf
>>> import matplotlib.pyplot as plt
>>>
>>> signal = pf.signals.impulse(64, 10)
>>>
>>> pf.plot.use()
>>> _, ax = plt.subplots(3, 1, figsize=(8, 8))
>>> pf.plot.time_freq(signal, ax=ax[:2], label="input", unit='ms')
>>> pf.plot.group_delay(signal, ax=ax[2], unit="samples")
>>>
>>> for order in [30, 6]:
>>> delayed = pf.dsp.fractional_time_shift(
... signal, 2.3, order=order)
>>> pf.plot.time_freq(delayed, ax=ax[:2],
... label=f"delayed, order={order}", unit='ms')
>>> pf.plot.group_delay(delayed, ax=ax[2], unit="samples")
>>>
>>> ax[1].set_ylim(-15, 5)
>>> ax[2].set_ylim(8, 14)
>>> ax[0].legend()
Apply a shift that exceeds the signal length using the modes ``"linear"``
and ``"cyclic"``
.. plot::
>>> import pyfar as pf
>>>
>>> signal = pf.signals.impulse(32, 16)
>>>
>>> ax = pf.plot.time(signal, label="input", unit='ms')
>>>
>>> for mode in ["cyclic", "linear"]:
>>> delayed = pf.dsp.fractional_time_shift(
... signal, 25.3, order=10, mode=mode)
>>> pf.plot.time(delayed, label=f"delayed, mode={mode}", unit='ms')
>>>
>>> ax.legend()
"""
# check input -------------------------------------------------------------
if not isinstance(signal, (pf.Signal)):
raise TypeError("Input data has to be of type pyfar.Signal")
if order <= 0:
raise ValueError("The order must be > 0")
if side_lobe_suppression <= 0:
raise ValueError("The side lobe suppression must be > 0")
if mode not in ["linear", "cyclic"]:
raise ValueError(
f"The mode is '{mode}' but must be 'linear' or 'cyclic'")
if order + 1 > signal.n_samples:
raise ValueError((f"The order is {order} but must not exceed "
f"{signal.n_samples-1} (signal.n_samples-1)"))
if unit == 's':
shift = shift*signal.sampling_rate
elif unit != 'samples':
raise ValueError(
f"Unit is '{unit}' but has to be 'samples' or 's'.")
# separate integer and fractional shift -----------------------------------
delay_int = np.atleast_1d(shift).astype(int)
delay_frac = np.atleast_1d(shift - delay_int)
# force delay_frac >= 0 as required by Laakso et al. 1996 Eq. (2)
mask = delay_frac < 0
delay_int[mask] -= 1
delay_frac[mask] += 1
# compute the sinc functions (fractional shift filters) -------------------
# Laakso et al. 1996 Eq. (21) applied to the fractional part of the shift
# M_opt essentially sets the center of the sinc function in the FIR filter.
# NOTE: This is also the shift that is added when applying the fractional
# part of the shift and has thus to be accounted for when realizing
# delay_int
if order % 2:
M_opt = delay_frac.astype("int") - (order-1)/2
else:
M_opt = np.round(delay_frac) - order / 2
# get matrix versions of the fractional shift and M_opt
delay_frac_matrix = np.tile(
delay_frac[..., np.newaxis],
tuple(np.ones(delay_frac.ndim, dtype="int")) + (order + 1, ))
M_opt_matrix = np.tile(
M_opt[..., np.newaxis],
tuple(np.ones(M_opt.ndim, dtype="int")) + (order + 1, ))
# discrete time vector
n = np.arange(order + 1) + M_opt_matrix - delay_frac_matrix
sinc = np.sinc(n)
# get the Kaiser windows --------------------------------------------------
# (dsp.time_window can not be used because we need to evaluate the window
# for non integer values)
# beta parameter for side lobe rejection according to
# Oppenheim (2010) Eq. (10.13)
beta = pf.dsp.kaiser_window_beta(abs(side_lobe_suppression))
# Kaiser window according to Oppenheim (2010) Eq. (10.12)
alpha = order / 2
L = np.arange(order + 1).astype("float") - delay_frac_matrix
# required to counter operations on M_opt and make sure that the maxima
# of the underlying continuous sinc function and Kaiser window appear at
# the same time
if order % 2:
L += .5
else:
L[delay_frac_matrix > .5] += 1
Z = beta * np.sqrt(np.array(1 - ((L - alpha) / alpha)**2, dtype="complex"))
# suppress small imaginary parts
kaiser = np.real(bessel_first_mod(0, Z)) / bessel_first_mod(0, beta)
# apply fractional shift --------------------------------------------------
# compute filter and match dimensions
frac_delay_filter = sinc * kaiser
while frac_delay_filter.ndim < signal.time.ndim:
frac_delay_filter = frac_delay_filter[np.newaxis]
# apply filter
convolve_mode = mode if mode == "cyclic" else "full"
n_samples = signal.n_samples
signal = pf.dsp.convolve(
signal, pf.Signal(frac_delay_filter, signal.sampling_rate),
mode=convolve_mode)
# apply integer shift -----------------------------------------------------
# account for shift from applying the fractional filter
delay_int += M_opt.astype("int")
signal = pf.dsp.time_shift(signal, delay_int, mode)
# truncate signal (got padded during convolution with mode='full')
if mode == "linear":
signal.time = signal.time[..., :n_samples]
return signal
[docs]
def resample(signal, sampling_rate, match_amplitude="auto", frac_limit=None,
post_filter=False):
"""Resample signal to new sampling rate.
The SciPy function ``scipy.signal.resample_poly`` is used for resampling.
The resampling ratio ``L = sampling_rate/signal.sampling_rate``
is approximated by a fraction of two integer numbers `up/down` to first
upsample the signal by `up` and then downsample by `down`. This way `up`
and `down` are smaller than the respective new and old sampling rates.
.. note ::
`sampling_rate` should be divisible by 10, otherwise it can cause an
infinite loop in the ``resample_poly`` function.
The amplitudes of the resampled signal can match the amplitude of the
input signal in the time or frequency domain. See the parameter
`match_amplitude` and the examples for more information.
Parameters
----------
signal : Signal
Input data to be resampled
sampling_rate : number
The new sampling rate in Hz
match_amplitude : string
Define the domain to match the amplitude of the resampled data.
``'auto'``
Chooses domain to maintain the amplitude automatically, depending
on the ``signal.signal_type``. Sets ``match_amplitude == 'freq'``
for ``signal.signal_type = 'energy'`` like impulse responses and
``match_amplitude == 'time'`` for other signals.
``'time'``
Maintains the amplitude in the time domain. This is useful for
recordings such as speech or music and must be used if
``signal.signal_type = 'power'``.
``'freq'``
Maintains the amplitude in the frequency domain by multiplying the
resampled signal by ``1/L`` (see above). This is often desired
when resampling impulse responses.
The default is ``'auto'``.
frac_limit : int
Limit the denominator for approximating the resampling factor `L`
(see above). This can be used in case the resampling gets stuck in an
infinite loop (see note above) at the potenital cost of not exactly
realizing the target sampling rate.
The default is ``None``, which uses ``frac_limit = 1e6``.
post_filter : bool, optional
In some cases the up-sampling causes artifacts above the Nyquist
frequency of the input signal, i.e., ``signal.sampling_rate/2``. If
``True`` the artifacts are suppressed by applying a zero-phase Elliptic
filter with a pass band ripple of 0.1 dB, a stop band attenuation of 60
dB. The pass band edge frequency is ``signal.sampling_rate/2``. The
stop band edge frequency is the minimum of 1.05 times the pass band
frequency and the new Nyquist frequency (``sampling_rate/2``). The
default is ``False``. Note that this is only applied in case of
up-sampling.
Returns
-------
signal : pyfar.Signal
The resampled signal of the input data with a length of
`up/down * signal.n_samples` samples.
Examples
--------
For power signals, the amplitude of the resampled signal is automatically
correct in the time `and` frequency domain if ``match_amplitude="time"``
.. plot::
>>> import pyfar as pf
>>> import matplotlib.pyplot as plt
>>>
>>> signal = pf.signals.sine(200, 4800, sampling_rate=48000)
>>> resampled = pf.dsp.resample(signal, 96000)
>>>
>>> pf.plot.time_freq(signal, label="original", unit='ms')
>>> pf.plot.time_freq(resampled, c="y", ls=":", unit='ms',
... label="resampled (time domain matched)")
>>> plt.legend()
With some energy signals, such as impulse responses, the amplitude can only
be correct in the time `or` frequency domain due to the lack of
normalization by the number of samples. In such cases, it is often desired
to match the amplitude in the frequency domain
.. plot::
>>> import pyfar as pf
>>> import matplotlib.pyplot as plt
>>>
>>> signal = pf.signals.impulse(128, 64, sampling_rate=48000)
>>> resampled_time = pf.dsp.resample(
... signal, 96000, match_amplitude = "time")
>>> resampled_freq = pf.dsp.resample(
... signal, 96000, match_amplitude = "freq")
>>>
>>> pf.plot.time_freq(signal, label="original", unit='ms')
>>> pf.plot.time_freq(resampled_freq, dashes=[2, 3], unit='ms',
... label="resampled (freq. domain matched)")
>>> ax = pf.plot.time_freq(resampled_time, ls=":", unit='ms',
... label="resampled (time domain matched)", c='y')
>>> ax[0].set_xlim(1.2,1.46)
>>> plt.legend()
"""
# check input
if not isinstance(signal, (pf.Signal)):
raise TypeError("Input data has to be of type pyfar.Signal")
# calculate factor L for up- or downsampling
sampling_rate_old = signal.sampling_rate
L = sampling_rate / sampling_rate_old
# set match_amplitude domain depending on signal.signal_type
if match_amplitude == "auto":
match_amplitude = "freq" if signal.signal_type == "energy" else "time"
# set gain depending on domain to match aplitude in
if match_amplitude == "time":
gain = 1
elif match_amplitude == "freq":
gain = 1/L
# the aplitude of signals with signal_type "power" must be matched in
# the time domain
if signal.signal_type == "power":
raise ValueError((
'match_amplitude must be "time" if signal.signal_type is '
'"power".'))
else:
raise ValueError((f"match_amplitude is '{match_amplitude}' but must be"
" 'auto', 'time' or 'freq'"))
# check if one of the sampling rates is not divisible by 10
if sampling_rate % 10 or sampling_rate_old % 10:
warnings.warn((
'At least one sampling rate is not divisible by 10, , which can '
'cause a infinite loop in `scipy.resample_poly`. If this occurs, '
'interrupt and choose different sampling rates or decrease '
'frac_limit. However, this can cause an error in the target '
'sampling rate realisation.'))
# give the numerator and denomitor of the fraction for factor L
if frac_limit is None:
frac = Fraction(Decimal(L)).limit_denominator()
else:
frac = Fraction(Decimal(L)).limit_denominator(frac_limit)
up, down = frac.numerator, frac.denominator
# calculate an error depending on samplings rates and fraction
error = abs(sampling_rate_old * up / down - sampling_rate)
if error != 0.0:
warnings.warn((
f'The target sampling rate was realized with an error of {error}.'
f'The error might be decreased by setting `frac_limit` to a value '
f'larger than {down} (This warning is not shown, if the target '
'sampling rate can exactly be realized).'))
# resample data with scipy resampe_poly function
data = sgn.resample_poly(signal.time, up, down, axis=-1)
data = pf.Signal(data * gain, sampling_rate, fft_norm=signal.fft_norm,
comment=signal.comment)
if post_filter and L > 1:
# Design elliptic filter
# (pass band is given by nyquist frequency of input signal, other
# parameters are freely chosen)
wp = sampling_rate_old / 2 / sampling_rate * 2
ws = min(1, 1.05 * wp)
gpass = .1
gstop = 60
# calculate the required order and -3 dB cut-off frequency
N, f_c = sgn.ellipord(wp, ws, gpass, gstop/2, fs=sampling_rate)
f_c *= sampling_rate / 2
# apply zero-phase filter
data = pf.dsp.filter.elliptic(data, N, gpass, gstop/2, f_c, 'lowpass')
data.time = np.flip(data.time, axis=-1)
data = pf.dsp.filter.elliptic(data, N, gpass, gstop/2, f_c, 'lowpass')
data.time = np.flip(data.time, axis=-1)
return data
[docs]
class InterpolateSpectrum():
"""
Interpolate an incomplete spectrum to a complete single sided spectrum.
This is intended to interpolate transfer functions, for example sparse
spectra that are defined only at octave frequencies or incomplete spectra
from numerical simulations.
Parameters
----------
data : FrequencyData
Input data to be interpolated.
method : string
Specifies the input data for the interpolation
``'complex'``
Separate interpolation of the real and imaginary part
``'magnitude_phase'``
Separate interpolation if the magnitude and unwrapped phase values
``'magnitude'``
Interpolate the magnitude values only. Results in a zero phase
signal, which is symmetric around the first sample. This phase
response might not be ideal for many applications. Minimum and
linear phase responses can be generated with
:py:func:`~pyfar.dsp.minimum_phase` and
:py:func:`~pyfar.dsp.linear_phase`.
kind : tuple
Three element tuple ``('first', 'second', 'third')`` that specifies the
kind of inter/extrapolation below the lowest frequency (first), between
the lowest and highest frequency (second), and above the highest
frequency (third).
The individual strings have to be
``'zero'``, ``slinear``, ``'quadratic'``, ``'cubic'``
Spline interpolation of zeroth, first, second or third order
``'previous'``, ``'next'``
Simply return the previous or next value of the point
``'nearest-up'``, ``'nearest'``
Differ when interpolating half-integers (e.g. 0.5, 1.5) in that
``'nearest-up'`` rounds up and ``'nearest'`` rounds down.
The interpolation is done using ``scipy.interpolate.interp1d``.
fscale : string, optional
``'linear'``
Interpolate on a linear frequency axis.
``'log'``
Interpolate on a logarithmic frequency axis.
The default is ``'linear'``.
clip : bool, tuple
The interpolated magnitude response is clipped to the range specified
by this two element tuple. E.g., ``clip=(0, 1)`` will assure that no
values smaller than 0 and larger than 1 occur in the interpolated
magnitude response. The clipping is applied after the interpolation.
The default is ``False`` which does not clip the data.
Returns
-------
interpolator : :py:class:`InterpolateSpectrum`
The interpolator can be called to interpolate the data (see examples
below). It returns a :py:class:`~pyfar.classes.audio.Signal` and has
the following parameters
`n_samples` : int
Length of the interpolated time signal in samples
`sampling_rate`: int
Sampling rate of the output signal in Hz
`show` : bool, optional
Show a plot of the input and output data. The default is ``False``.
Examples
--------
Interpolate a magnitude spectrum, add an artificial linear phase and
inspect the results.
Note that a similar plot can be created by the interpolator object by
``signal = interpolator(64, 44100, show=True)``
.. plot::
>>> import pyfar as pf
>>> import matplotlib.pyplot as plt
>>> import numpy as np
>>>
>>> pf.plot.use()
>>> _, ax = plt.subplots(2, 3)
>>>
>>> # generate data
>>> data = pf.FrequencyData([1, 0], [5e3, 20e3])
>>>
>>> # interpolate and plot
>>> for ff, fscale in enumerate(["linear", "log"]):
>>> interpolator = pf.dsp.InterpolateSpectrum(
... data, 'magnitude', ('nearest', 'linear', 'nearest'),
... fscale)
>>>
>>> # interpolate to 64 samples linear phase impulse response
>>> signal = interpolator(64, 44100)
>>> signal = pf.dsp.linear_phase(signal, 32)
>>>
>>> # time signal (linear and logarithmic amplitude)
>>> pf.plot.time(signal, ax=ax[ff, 0], unit='ms', dB=True)
>>> # frequency plot (linear x-axis)
>>> pf.plot.freq(
... signal, dB=False, freq_scale="linear", ax=ax[ff, 1])
>>> pf.plot.freq(data, dB=False, freq_scale="linear",
... ax=ax[ff, 1], c='r', ls='', marker='.')
>>> ax[ff, 1].set_xlim(0, signal.sampling_rate/2)
>>> ax[ff, 1].set_title(
... f"Interpolated on {fscale} frequency scale")
>>> # frequency plot (log x-axis)
>>> pf.plot.freq(signal, dB=False, ax=ax[ff, 2], label='input')
>>> pf.plot.freq(data, dB=False, ax=ax[ff, 2],
... c='r', ls='', marker='.', label='output')
>>> ax[ff, 2].set_xlim(2e3, signal.sampling_rate/2)
>>> ax[ff, 2].legend(loc='best')
"""
def __init__(self, data, method, kind, fscale='linear', clip=False):
# check input ---------------------------------------------------------
# ... data
if not isinstance(data, pf.FrequencyData):
raise TypeError('data must be a FrequencyData object.')
if data.n_bins < 2:
raise ValueError("data.n_bins must be at least 2")
# ... method
methods = ['complex', 'magnitude_phase', 'magnitude']
if method not in methods:
raise ValueError((f"method is '{method}'' but must be on of the "
f"following: {', '.join(methods)}"))
# ... kind
if not isinstance(kind, tuple) or len(kind) != 3:
raise ValueError("kind must be a tuple of length 3")
kinds = ['linear', 'nearest', 'nearest-up', 'zero', 'slinear',
'quadratic', 'cubic', 'previous', 'next']
for k in kind:
if k not in kinds:
raise ValueError((f"kind contains '{k}' but must only contain "
f"the following: {', '.join(kinds)}"))
# ... fscale
if fscale not in ["linear", "log"]:
raise ValueError(
f"fscale is '{fscale}'' but must be linear or log")
# ... clip
if clip:
if not isinstance(clip, tuple) or len(clip) != 2:
raise ValueError("clip must be a tuple of length 2")
# initialize the interpolators ----------------------------------------
# store required parameters
self._method = method
self._clip = clip
self._fscale = fscale
self._kind = kind
# flatten input data to work with scipy interpolators
self._cshape = data.cshape
data = data.flatten()
self._input = data
# get the required data for interpolation
if method == 'complex':
self._data = [np.real(data.freq), np.imag(data.freq)]
elif method == 'magnitude_phase':
self._data = [np.abs(data.freq),
pf.dsp.phase(data, unwrap=True)]
else:
self._data = [np.abs(data.freq)]
# frequencies for interpolation (store for testing)
self._f_in = data.frequencies.copy()
def __call__(self, n_samples, sampling_rate, show=False):
"""
Interpolate a Signal with n_samples length.
(see class docstring) for more information.
"""
# length of half sided spectrum and highest frequency
n_fft = n_samples//2 + 1
f_max = sampling_rate / n_samples * (n_fft - 1)
# get the frequency values
if self._fscale == "linear":
# linearly spaced frequencies
self._f_query = pf.dsp.fft.rfftfreq(n_samples, sampling_rate)
self._f_base = self._f_in
else:
# logarithmically scaled frequencies between 0 and log10(n_fft)
self._f_query = np.log10(np.arange(1, n_fft+1))
self._f_base = np.log10(self._f_in / f_max * (n_fft - 1) + 1)
# frequency range
self._freq_range = [self._f_base[0], self._f_base[-1]]
# get the interpolators
self._interpolators = []
for d in self._data:
interpolators = []
for idx, k in enumerate(self._kind):
if idx == 1:
interpolators.append(interp1d(self._f_base, d, k))
else:
interpolators.append(interp1d(
self._f_base, d, k, fill_value="extrapolate"))
self._interpolators.append(interpolators)
# get interpolation ranges
id_below = self._f_query < self._freq_range[0]
id_within = np.logical_and(self._f_query >= self._freq_range[0],
self._f_query <= self._freq_range[1])
id_above = self._f_query > self._freq_range[1]
# interpolate the data
interpolated = []
for data in self._interpolators:
data_interpolated = np.concatenate((
(data[0](self._f_query[id_below])),
(data[1](self._f_query[id_within])),
(data[2](self._f_query[id_above]))),
axis=-1)
interpolated.append(data_interpolated)
# get half sided spectrum
if self._method == "complex":
freq = interpolated[0] + 1j * interpolated[1]
elif self._method == 'magnitude_phase':
freq = interpolated[0] * np.exp(-1j * interpolated[1])
else:
freq = interpolated[0]
# get initial signal
signal = pf.Signal(freq, sampling_rate, n_samples, "freq")
# clip the magnitude
if self._clip:
signal.freq = np.clip(
np.abs(signal.freq),
self._clip[0],
self._clip[1]) * np.exp(-1j * pf.dsp.phase(signal))
if show:
# plot input and output data
with pf.plot.context():
_, ax = plt.subplots(2, 2)
# time signal (linear amplitude)
pf.plot.time(signal, ax=ax[0, 0])
# time signal (log amplitude)
pf.plot.time(signal, ax=ax[1, 0], dB=True)
# frequency plot (linear x-axis)
pf.plot.freq(signal, dB=False, freq_scale="linear",
ax=ax[0, 1])
pf.plot.freq(self._input, dB=False, freq_scale="linear",
ax=ax[0, 1], c='r', ls='', marker='.')
ax[0, 1].set_xlim(0, sampling_rate/2)
# frequency plot (log x-axis)
pf.plot.freq(signal, dB=False, ax=ax[1, 1], label='output')
pf.plot.freq(self._input, dB=False, ax=ax[1, 1],
c='r', ls='', marker='.', label='intput')
min_freq = np.min([sampling_rate / n_samples,
self._input.frequencies[0]])
ax[1, 1].set_xlim(min_freq, sampling_rate/2)
ax[1, 1].legend(loc='best')
return signal
