Page 508 - Advanced engineering mathematics
P. 508

488    CHAPTER 14  The Fourier Integral and Transforms

                                    Now recall time and frequency convolution for the Fourier transform. Analog filtering in the
                                 time variable is done by convolution. If ϕ(t) is the filter function, then the effect of filtering f
                                 by ϕ is
                                                                        ∞

                                                      f ϕ (t) = (ϕ ∗ f )(t) =  ϕ(τ) f (t − τ)dτ.
                                                                        −∞
                                 Take the Fourier transform of this equation to obtain
                                                                ˆ
                                                               f ϕ (ω) =ˆϕ(ω) f (ω).
                                                                           ˆ
                                 We therefore filter f in the frequency variable by taking a product of the Fourier transform of
                                 the filter function ϕ with the transform of f .
                                    Using the convolution theorem, we can formulate equation (14.15) as

                                                                    sin(ω 0 t)
                                                              (t) =        ∗ f (t) .
                                                            f ω 0
                                                                      πt
                                 This gives the low-pass filtering of f as the convolution of the Shannon sampling function
                                 with f .
                                                                                   from f . This filters out frequen-
                                    In low-pass filtering, we produce a band-limited signal f ω 0
                                 cies outside of [−ω 0 ,ω 0 ]. In a similar kind of filtering, called bandpass filtering, we want to filter
                                 out the effects of the signal outside of given bandwidths. A band-limited signal f can always be
                                 decomposed into a sum of signals, each of which carries the information content of f within a
                                 specified frequency band. To see how to do this suppose f is band-limited with bandwidth 
.
                                 Consider a finite increasing sequence of frequencies

                                                           0 <ω 1 <ω 2 < ··· <ω N = 
.
                                 For j = 1,2,··· , N, define a bandwidth filter function β j by means of its transfer function:
                                                            ˆ β j = χ [−ω j ,−ω j−1 ] + χ [ω j−1 ,ω j ] .

                                 This is a sum of characteristic functions of frequency intervals, and is zero outside of these
                                 intervals and 1 for −ω j ≤ ω ≤−ω j−1 and ω j−1 ≤ ω ≤ ω j . The bandwidth filter function β j (t),
                                 which filters the frequency content of f (t) outside of the frequency range [ω j−1 ,ω j ], is obtained
                                 as the inverse Fourier transform of ˆ β j (ω). We obtain
                                                                 sin(ω j t) − sin(ω j−1 t)
                                                           β j (t) =              .
                                                                         πt
                                    A graph of this function is shown in Figure 14.6. Now define functions
                                                                    sin(ω 0 t)
                                                              f 0 (t) =    ∗ f (t)
                                                                      πt
                                 and, for j = 1,2,··· , N,
                                                               f j (t) = β j (t) ∗ f (t).

                                 Then, for j =1,2,··· , N, f j (t) carries the content of f (t) in the frequency range ω j−1 ≤ω ≤ω j ,
                                 while f 0 (t) carries the content in [0,ω 0 ], the low-frequency range of f (t). Furthermore,
                                                         f (t) = f 0 (t) + f 1 (t) + ··· + f N (t).
                                 This is a decomposition of the signal into components, each carrying information about the
                                 frequency in a specific frequency interval.
                                    We conclude this section with an observation connecting low-pass filters to a previously seen
                                 phenomenon. In signal analysis, the Gibbs phenomenon can be thought of as the step response of
                                 a low-pass filter. In this context the oscillations near the point of discontinuity are called ringing
                                 artifacts.




                      Copyright 2010 Cengage Learning. All Rights Reserved. May not be copied, scanned, or duplicated, in whole or in part. Due to electronic rights, some third party content may be suppressed from the eBook and/or eChapter(s).
                      Editorial review has deemed that any suppressed content does not materially affect the overall learning experience. Cengage Learning reserves the right to remove additional content at any time if subsequent rights restrictions require it.

                                   October 14, 2010  16:43  THM/NEIL   Page-488        27410_14_ch14_p465-504
   503   504   505   506   507   508   509   510   511   512   513