NOTE: THIS DOCUMENT IS OBSOLETE, PLEASE CHECK THE NEW VERSION: "Mathematics of the Discrete Fourier Transform (DFT), with Audio Applications --- Second Edition", by Julius O. Smith III, W3K Publishing, 2007, ISBN 978-0-9745607-4-8. - Copyright © 2017-09-28 by Julius O. Smith III - Center for Computer Research in Music and Acoustics (CCRMA), Stanford University
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Beginning with Eq. (B.4.3), we have
where is the transform of the filter input, is the transform of the output signal, and is the filter transfer function.
Definition. The frequency response of a linear time-invariant digital filter is defined to be the transfer function, , evaluated on the unit circle, that is, .
The frequency response is a complex-valued function of a real variable. The response at frequency Hz, for example, is , where is the sampling period in seconds.
Since every complex number can be represented as a magnitude and angle, the frequency response may be decomposed into two real-valued functions, the amplitude response and the phase response. Formally, we may define them as follows:
Thus is the magnitude (or complex modulus) of , and is the phase (or complex angle) of .
Definition. The real valued function in Eq. (B.21) is called the filter amplitude response and it specifies the amplitude gain that the filter provides at each frequency.
Definition. The function is called the power response and it specifies the power gain at each frequency.
Definition. The real function in Eq. (B.21) is thephase response and it gives the phase shift in radians that each input component sinusoid will undergo.
If the filter input and output signals are and respectively, then