mindspore.dataset.audio.LFilter

class mindspore.dataset.audio.LFilter(a_coeffs, b_coeffs, clamp=True)[源代码]

Design two-pole filter for audio waveform of dimension of (…, time).

Parameters
  • a_coeffs (sequence) – denominator coefficients of difference equation of dimension of (n_order + 1). Lower delays coefficients are first, e.g. [a0, a1, a2, …]. Must be same size as b_coeffs (pad with 0’s as necessary).

  • b_coeffs (sequence) – numerator coefficients of difference equation of dimension of (n_order + 1). Lower delays coefficients are first, e.g. [b0, b1, b2, …]. Must be same size as a_coeffs (pad with 0’s as necessary).

  • clamp (bool, optional) – If True, clamp the output signal to be in the range [-1, 1] (default=True).

Examples

>>> import numpy as np
>>>
>>> waveform = np.array([[2.716064453125e-03, 6.34765625e-03], [9.246826171875e-03, 1.0894775390625e-02]])
>>> a_coeffs = [0.1, 0.2, 0.3]
>>> b_coeffs = [0.1, 0.2, 0.3]
>>> numpy_slices_dataset = ds.NumpySlicesDataset(data=waveform, column_names=["audio"])
>>> transforms = [audio.LFilter(a_coeffs, b_coeffs)]
>>> numpy_slices_dataset = numpy_slices_dataset.map(operations=transforms, input_columns=["audio"])