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In this Chapter will present the transition from the Z transform to the DFT (Discrete Fourier Transform). The DFT is used to compute the Fourier transform of discrete data.

3.1

The Z transform and the DFT

We have already deﬁned the Z-transform of a time series as follows:

The Z-transform provides a representation of our time series in terms of a polynomial. Let us introduce the following deﬁnition:

in this case the z-transform becomes, the DFT:

is a complex variable We have mapped the Z-transform into the unit circle ( of unit magnitude and phase given by ). The phase is also the frequency variable in 61

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(3.1)

(3.2)

(3.3)

So far. . The DFT maps a discrete signal into the frequency domain. . .5) (3.4) ¨© & & " 4" © e ih¨© qgIdcX ' % © e ih3© dcX ' % B " b X g © &' © ' B VVW `` X & X © feIdcX &' % B VVVV § ```` ¤ e %dcX % UVVV a```Y § s qu" t§ r p & © ae9cdX ' % efCcdB X % X & ( ' % 4 B 0 ©& ( ' % ¥ T10¢ 5 ¡ UVVV VVV VVVW . It is easy to make an analogy with the Fourier transform when the time is measured In the last equation the frequency is given in in seconds. Since does not have units. . the frequency is a continuous variable. . . . In the DFT the integration is replaced by nel summation and the Fourier Kernel by .6) § S 5 (3. . remember that is an angular frequency. . . The limits of are given by . DISCRETE FOURIER TRANSFORM units of radians. .8) 0 ¨© © ih¨© ih¨© ' & b b e g dcX % g & ``` X VVVW " ``` © e ih ¨© qgIdcX & ' % X ``` VVV ``` © e ih3B© dcX ' % g a```Y UVVV a```Y 0 0 0 B E 4 " )B) H& § ¥5 " RQP) 4" " #IA' § 7 ) )(' ¥% ¡ ¨ © ¨46 ¡0 4 B E "" )B) GFD4) 44" #CA' § 7 @981 ¢§ 60 ) 4 3 7 5 4 " $%% #! ¥ © & ¦£ £ ¡ ¥ § )(' ¥% ¨§¡ ¤¢§ 210¡ " 4" " " 4" " " 4" " (3. . . . We can also write down our transform in matrix form . In the Fourier transform an analog signal is multiplied by the Fourier kerand integrated with respect to time. . but let us assume that one wants to discretized in the same way we have discretized the temporal variable . .7) (3. . has units of radians. . The last equation can be written in compact form as follows: 31) ¥ 20('& (3.62 CHAPTER 3. we can discretized the frequency axis as follows: Now we can deﬁne the DFT as follows: Now the DFT is a transformation of a points signal into Fourier coefﬁcients . . . If the time series is a points time series.

ﬁrst we replace the last equation into equation (3. however it is important to note that because the discrete nature of the problem we have interchanged the integration symbol by a summation.3. in other words we need 3. We need a transform to come back from the frequency domain to the time . The parameters are our unknowns. The remaining problem entails the invertivility of the DFT.1 Inverse DFT We propose the following inverse where the coefﬁcients must be determined.1. The last equation can be rewritten as At this point we realize the the last equation is a geometric series1 with a sum given by 1 We have used a geometric series to ﬁnd the inverse of a dipole in Chapter 2 " © e h 5 g dcX & % ¨ © § 5 " ¡ ¡ ¡ where the sequence ) 5 " ¨ © § © e h 5 g dcX & % ¨ © ¨ © § ¤ 5 is given by ¡ ¡ ¢ 5 " © e h 5 g dcX & % ¨ © ¨ © § ¤ ¡ " © e dcX & % 3 © § ¡ ¡ ¡ ¡ ¡ ¢ ¢ ¡ ¡ s ¡ ¢ ¡ 5 " ¢ 5 ¢ (3.9) (3.12) . domain.11) (3. This formula is analogous to the one use to invert the Fourier transform.10) (3.1.6). THE Z TRANSFORM AND THE DFT 63 It is clear that the DFT can be interpreted as a matrix that maps an -dimensional vector into another -dimensional vector. In order to ﬁnd our unknowns we proceed as follows.

18) where is an identity matrix.17) 4 4 s © s .16) (3. the DFT and the IDFT (inverse DFT) .11) we obtain the following expression for our unknown coefﬁcients our inversion formula becomes: This equation can also be written as follows: " r s B § u (3. ) BGEF4D) & " 4" A' § 7 ) © e CdcX ' % ¨ © § 5 " ) 5 B E 4 "" ) 5 " GFD) 44" A' § 7 ) 5 4 § ¤ s ) 7¢§ § © 7 ¢§ © 4 " © e dcX & % 3 © B § ¡ ) © 4 § s s ¡ 4 ¡ ' 4 ¡ £¢ ¢ §5 " In equation (3.64 CHAPTER 3. It is clear that the ©h & e 5 g dcX % § ¤ ¥ ¦¤ 4 ¡ £¢ 4 § 3 © ¡ (3.13) (3.19) 4 4 The matrix is the Hermitian transpose of the matrix matrix is an orthogonal matrix. DISCRETE FOURIER TRANSFORM if if ¡ if if after introducing the ﬁnal result into equation (3.12) we can identify . Finally we have a pair of transforms. therefore B §§ B ¨ § . given by (3.15) (3.14) 5 ¢ (3.

the IDFT is used to map a signal in the frequency domain into time domain. The sampling interval of the frequency axis is now In general. It is also important to pad with zeros at the time of performing discrete convolution using the DFT. 3. the cost of multiplying a matrix by a vector is proportional to . in this case we deﬁne a new time series that condecrease the frequency interval sists of the original time series followed by zeros. The cost of inverting an matrix is proportional to . where the index is associated to the discrete frequency .3. the frequency axis is sampled every radians. Zero padding can be used to . We will further diminish the computation cost of multiplying a matrix times a vector by using the FFT (Fast Fourier Transform). the inverse is computed using the Hermitian operator . Because. the trick of zero padding is used to oversample the frequency axis at the time of plotting the DFT.1. B E ) & 4 & 5 " ) 4" " A' § 7 ) © D5 dcX ' % © § © D5 dcX ' % ¨ © § ¤ e e 3 " 1 4 3 § 0 1 ¢$¢ The new time series is an -points time series with a DFT given by 0 $¢ 4 B E 4 " ) " GFD) " 4" A' § © e IdcX & % ¤ ¨ 5 © B § 5 5 (3.2 Zero padding The DFT allows us to transform an -points time series into frequency coefﬁcients .20) 45 0 0 ¢ 4 0 $¢ 4 BGEF4D) " " ) B ) 0 ¢ 5 44" #IA' § 7 ) 7 $£§ 7 3 1 § 60 ¡4 ¨ ' §¦¥ ) © ' ) ¤ 3© ) ) ) ) & § " " 4" X " D) " 4" AD0' © 4 E X 4 4 4 4 7 5 ¤ (3.21) . THE Z TRANSFORM AND THE DFT 65 The DFT is used to map a discrete signal into the frequency domain. the DFT is an orthogonal transformation. At this point it appears that is controlled by the number of samples of the time series .1.

In Figure (3. axis is given in radians In Figures (3.5 2 Imag[Xk] 1 0 0 1 2 3 ω [rad] 4 5 6 7 −1 −2 0 1 2 3 4 5 6 7 ω [rad] Figure 3.1) and (3.2).2 x 0 −0. This codes was utilized to generate Figures (3.2) we portray the effect of padding a time series. In the following example I show how to pad with zeros a time series. In Figure (3. DISCRETE FOURIER TRANSFORM 0.4 0.2 1 Real[Xk] 0. d GF £¥©£¥b a@"c@¦¨A4 )¨©T(B¨¦9AQ¦¨&$ )0 § S§¥£R E§ ¥£P I GF £CB E§ CB£8 @4¨)4AD))A@9 5 cu¢&¨¦¢)i4¦¨4 v8 E S§¥£R CB b ' 1 6 e CB£ 1 Wr #))@A¦' 20 6ur 8 U W&¨¦@# y1 # w1 cu eS§¥£R x 6U v8 E 6 %tsf# 1 # d 6W¦ r r dU 5 g ¦U ¨qW)pihgd H f&S 1 # e g © e 6 V VU a`0YXUWQ1 65H 7¤91 8 653 1 7&420 £'©$#! © §¥£¡ )(&%" ¤¦¨¨¦¤¢ 1) ¥ 3 2A' ¡ . Note that freq.66 CHAPTER 3.2) the original time series after zero padding ( zeros) is used to compute the DFT.1) and (3.1) we have the original time series and the associated DFT (the real and imaginary part).5 0 0 5 10 15 n 20 25 30 −0.1: A time series and the real and imaginary parts of the DFT.

THE Z TRANSFORM AND THE DFT The FFT is not a new transform. 0 ¢ 3.3 3.1.1.5 −0.r © &$" x ¦ %&Y£ A¦' Yr © ¦@¥ x ¦£(i¤e '¨A¦' # W¤ e ¦ X fA@uS #! $ ©e ' P 6 $§ © £P 6 rr $ e $£B 6 3U e B§'© Wr i3 a%¤(&P I S r © &#$" x '¤¨eY£ A¦' 6Yr © ¦@¥ x ¦(i§¤e ¨PA¦' # Wr¤ Y¤@¥ fA@uS ! £ © ' P £$ © '£ 6 r e'£ e $£B 6 rHU e B§'© W¨u3 a%¤(&P I S 6 © © '£ 6 © e P ' 6 e B§' Wr #eY¨P ¦' Wr `©Y'£ A¦¨# Wr #fqaA¤(© 6 UU e B§'© Wr¦¤3 a%¤(&P I S SB S£ B§'b (A' I @¥2%¤(4 6 sg© H 0 (i)pig 91 d (¦¥¢c¤(¦ d¨&E ¨¤B ¨B I &$i¨¢ S S # §£¥ £C £ © §¡ 6Wr #)¤E 2 F¥£ £ £ ©$§ )¦¤¨C¤B ¨B I &i¨¡¢ e BE 1 6 A£ 6 V VU a`0YXUWQ1 Figure 3. the FFT is just a fast algorithm to compute DFTs. that is a trick to compute the DFT of length N time series using the DFT of two sub-series of length N/2. The FFT is based on the halving trick.5 0.2 0. In this case the time series was padded with zeros in order to decrease the frequency interval .2 0.2: A time series and the real and imaginary parts of the DFT. Let’s start assuming that we a have : a time series of length 41 The Fast Fourier Transform (FFT) −0.4 −2 −1 Imag[Xk] −1 0 1 2 Real[Xk] 0 1 0 x 0 0 0 5 1 1 10 2 2 15 20 3 3 ω [rad] ω [rad] 25 n 4 4 30 35 5 5 40 6 6 45 50 7 7 67 .

24) and (3.68 CHAPTER 3.27) £ £ 4 h ¦h e X & X " 6© gX eP5 hi ¨¥ X cdX &' % ¨© XI£ ¨ © §6© gX P5 IdcX ' % I£ 3 © § 5 g "" ¤ 44" ¥£ ) £ ) £ § £ "" p X £ 44" C£ ) I£ ) § E 1 "" h h & © ¨ " BGF4 D) 44" ) 4 § 7 ) 6© T© ' egX ¨ g dcX % £ ¨© X § 4 E7 § h & " B E 4 1D) "4" " ) 4 § 7 ) T© ' gX De5 dcX % £ ¨© X § 5 ) BGEF4 1 ¡¢' § 7 ) 6© ' h & gX De5 dcX % £ 3© X § 5 B E 4¡ h & 5 " RQ' § 7 ) T© gX DIdcX ' % ¦ ¤ § 5 e5 4 £ " 3© X " X I£ ) 4" " ) £ ) I£ ) £ ) £ £ B 4 (3. It is clear that the RHS term can be written in terms of the DFTs of (even samples) and (odd samples) samples of the DFT of The last equation provides a formula to compute the ﬁrst based on the samples of the DFT of and . we will assume that one wants to compute the DFT of the time series . (3. respectively.24) .23) 4 £ (3.27) are applied 41 B E 4) h & © " GFDA' § ) T© gX DIdcX ' % E § ¨ e5 £ After rewriting the last expression in terms of mula: and we end up with the following for- (3.26) as a function Now we have two expression to compute the DFT of a series of length of two time series of length .22) (3.25) In the last equation we apply the following substitution: (3. Using the deﬁnition we can rewrite the last equation in terms of two time series composed of even samples and odd samples . The recursions given in (3. Now. note that we need another formula to retrieve the second half of the samples of the DFT of . Good FFT algorithms repeat this trick until the ﬁnal time series are series of length . DISCRETE FOURIER TRANSFORM First.

This is an important saving with respect to the standard DFT which involves a number of .29) (3.3.4 Working with the DFT/FFT Symmetries the frequency in radians is given by Using the following property we can re-write equation (3.24) and (3.1. 3. THE Z TRANSFORM AND THE DFT 69 to recover the DFT of the original time series. operations proportional to A small modiﬁcation to formulas (3.28) (3. The time series is The DFT is given by 1) E) ©) ¦)B) ¦) D#B C0) §0¨#IA§0 &1 § & ¨© ¤5 " ¥ § © e h 5 ¨ © g dcX & % ¨ © § © e h 5 ¨© g dcX ' % ¨ © § 5 BGEF4D) & 4 5 "" ) 44" 0' § 7 ) © D5 dcX ' % 3 © § ¤ e B E 4 "" )B) ) 4 3 5 " GFD) 44" #CA' § 7 @C 7 1 § 60 4 Let us start with a the DFT of a real time series of length 4 © ' § © h ¨© & % & e IdcX % e 5 dcX 5 g " r &0 e # 0 1 ¥ 4 ¡ X £¢¡4 3UH U 5 !d Hc¡i(%U d 5 555 55 ¤¤¤5 d 3 5¤¤&5 d 3UH d H¤H d U i(%U f 555 ¤¤&5 d U X 4 (3.27) will permit us to compute the inverse DFT. It can be proved that the total number of operations of the FFT is proportional to (for a time series of length ).30) #! $" 3 H U 5 # ¦(%¤G £'©$ .28) as follows: The following example is used to illustrate the last point.1.

In the previous example we have Note that the central frequency is . 3 ¢ £0 3 3 § 60 31) ¥ 3 2A &' 0 3 $¢ E 1 § ¡ 0 1 B E 4D) 4"4" A' § ) " ) 47 3 0 1 ¢ § C 77 $£§ 5 5 0 0 7 5 4 3 0 C81 § $¢ H 3UH d ¤H d U ¡i(%U f 555 3 555 ¤¤¤5 d 9 ¤¤&5 d 3UH U 5 !d H i(%U d 5 555 d ¤¤&5 H 3UH U 5 !d c¡i(%U d 5 55 555 5¤¤¤5 d 3 ¤¤&5 d H 3UH d ¤H d U i(%U f 555 ¤¤&5 d U #! $¦ 3 § T0 p B ¦1 4 It is claear that the ﬁrst the DFT. H 3 H 3UH d ¤H d U ¡i(%U f 5 5 55 ¤5¤¤5 d 39 5¤¤&5 d 3UH U 5 !d H i(%U d 5 555 d ¤¤&5 7 !a" d H d uU f 5 ¤ d 3 H U U d 3 H ¤ " 3 d H 5 "d U " d 5 5 # £$ $! ¦(i§ " U H " 5 3 # . Let us deﬁne the sampling interval in frequency as . We have already In the previous example I compute the DFT. DISCRETE FOURIER TRANSFORM The frequency axis in terms of samples . samples are required to deﬁne the remaining samples of U or .70 CHAPTER 3. If fact the is the Nyquist frequency in rads. . Well this simple reﬂect the way we have discretized the unit circle when computing the DFT. What is the meaning of frequencies above ?. therefore. . mentioned that is related to angular frequency as follows: . the last frequency is almost This is because we have discretized the unit circle in the interval In does not make much sense to talk about frequencies above radians.

33) . Let us ﬁrst consider a 2D discrete signal (i.2. a map) The formulas for the forward and inverse DFT in the 2D case are given by 4 & & ¦ ¥ 5 " D) 44" A' § © t ) 44" A' § 7 ) © e dcX ' % © e ¥ IdcX ' % ©¥ ¨ © © § ¦ ¤ "" ) ) "" ) 5 ¥ ) & X h £ ¢ £ ( ¨ ¢ ¡( g ' ¥% X ) ¡ ¡ ¡ 3¥ 60 ) 10¡ § X X " 60 0 h B )B E 4 " ) ¦ " GE ) " 4" A' § ¨ #GFD) " 4" A' § ) §¥ " ) £ ¢ £ ( X ¨ ¢ ¤( g & ¥% 60 ) ¡0 ¡ 3 q) 3 E ¡ ¥ ¡ § G¡ ¨¥ X ) ¡ ¡ 31) ¥ DA('& (3. It is important to stress that for our signal processing applications we will be dealing with the 2D DFT (this is the discrete version of the FT).3.31) (3.3: Distribution of frequencies around the unit circle.32) (3.. one can image the 2D FT as a decomposition of a signal in terms of plane waves. 3.e. we can deﬁne the inversion formula Whereas the 1D FT is used to decomposed signals in a decomposition of sin and cos.2 The 2D DFT The 2D Fourier transform is deﬁned as follows: similarly. The DFT can be plotted as in the interval or in the interval. THE 2D DFT 71 ω3 ω4=π ω5 ω2 ω1 ω0 11 ω0 2 ω1 3 ω2 4 5 6 7 8 ω3 ω4=π ω5 ω6 ω7 ω7 ω6 28 7 6 1 2 ω1 3 ω2 4 5 2π−ω5 ω0 2π−ω6 2π−ω7 ω3 ω4=π Figure 3.

72 CHAPTER 3. Notice that in the 2D DFT we need to consider 2D symmetries. then you compute the DFT to rows of the previous result. This is very simple: you ﬁrst compute the DFT of all the columns of .. 2D DFT codes are just 1D FFT’s codes working on rows and columns. ¦¥ 4 ¥ 4 4 " ) ¦ " D) 4" " A' § © ) ) " 4" A' § 7 ) © e dcX & % © e ¥ IdcX & % §¥ ¨ © © B § ¦ 5 " ) 5 ¡ (3.34) ¡ . In fact. DISCRETE FOURIER TRANSFORM The 2D DFT is computed by calling two times the 1D DFT.e. The 2D DFT is important at the time of ﬁltering 2D images (i. seismic records). gravity maps.

then the ﬁltered signal is given by In general.3. In this section we will examine the problem of designing FIR (Finite Impulse Response) ﬁlters.3.38) . 3. We can use the inverse Fourier transform to ﬁnd an expression for . In the previous expression is the cutoff frequency. it is more convenient to design short ﬁlters in the time domain and applied them via convolution2 where the sequence is the Impulse Response of the ﬁlter with desired amplitude response . 2 note that we are working with continuous signals.3.37) 5 ¥ 2 ¡0 (3. ON THE DESIGN OF FINITE IMPULSE RESPONSE FILTERS 73 3.3 On the Design of Finite Impulse Response ﬁlters So far we have studied operators (ﬁlters) that are capable of collapsing a wavelet into a spike.35) (3. These ﬁlters are often called spiking ﬁlters or Wiener ﬁlters. It is clear that if the signal to be ﬁltered is called .1 Low Pass FIR ﬁlters In this case we want to design a ﬁlter that operates in the time domain with a amplitude spectrum with the following characteristics: We will assume that the ﬁlter phase is zero. These are ﬁlters that are used to eliminate undesired spectral components from our data.36) (3. This ﬁlter can be either applied in the frequency domain or in the time domain. ¥ Evaluating the last integral leads to the following expression for the ﬁlter ): ¥§¡ ¡ 0 ¨§¡ ¥ ¡ 210¢ © % ¨§ ¦ % ¡ £ ¢0 ¥0 ¤¢0 £ ¡ © ¥ 0 )(& % 210¡ ¥ ¥ ¡ §¥ 210¡ " 20 ¢ 3210¡ " § E B' ¦ ( ( ¥ § ¡ ) ¥ ¨§¡ 3¥ § ¡ £ § 3 1 ¡ B § ¥§¡ ¡ £¢ §¥ 3210¡ (3.

We need to discretized the previous expression to obtain the impulse response of a discrete system: the factor comes from equation (1. We also display the associated amplitude response. One way to minimize truncation artifacts is by smoothing the truncated impulse response with a taper or window. The taper is used to minimize truncation effects at the end point of the impulse response. This point has already been studied in section (1.41) (3. the Fourier transform of the discrete and continuous signals are equal in The ﬁnal expression of the digital ﬁlter is given by It is clear that this is a IIR ﬁlter (inﬁnite impulse response ﬁlter). DISCRETE FOURIER TRANSFORM This is the impulse response of the continues system with amplitude response .3) where we examined the spectral artifacts that are introduced when a signal is truncated in time.40) B ¦ ¨ 1 B© B1 B ¦ ¨ 1 §¢ (3.2.4) we display the impulse response of a ﬁlter of cut-off frequency ter lengths ( ) . When the ﬁlter is truncated the actual amplitude spectrum of the ﬁlter is not equal to the desired or ideal amplitude spectrum (3. this is a scaling factor that allows us to say that . a popular taper is the Hamming Window: 3 § ¢ q) § ¢ 83 &E ¥ 210¡ ¥ £ ' § ¡ 3 ¦ ¢ ¡ " ¨ 44" ¨0D#IAD) B I) 1 I0¦ I) 44" ©E § ) ¥ § § ¢ ¡0 $ " ¢0 § £§ "" ¦) 1)B) ' E E) E "" ¨ ¡ 3 ¢ ¡ " 44" ¨0D#IAD) B I) 1 I0¦ I) 44" § ) ¥ § ¢ ¢ ¡0 $ " ¢0 § £§ "" ¦) 1)B) ' E E) E "" ¡ " ¡ § ¡ 3 § 3 ) ¥ ¢ ¡0 $ " ¢0 § ¥ ¡ 10¢¡¡ 0 " ¢0 § ¥ ¨§¡ §¡ ¡ ¡ § © ¥ © ¤ ¢ £ (3.35). and .39) ¥§¡ § ¢ § (3. © " § now is the truncated impulse response after applying a taper function. It is easy to see that the ﬁlter truncation has introduced the so called Gibbs phenomenon (Oscillations).44). In Figure for ﬁl(3.74 CHAPTER 3.42) . A FIR ﬁlter is obtained by truncating the IIR ﬁlter: In this case we have a ﬁlter of length .

4 0. It is important to stress that tapering will also increase the width of the transition band.1 0 −0.1 0 −0.5 Amplitude Response 0.2 1 0.4 0.1 −20 −10 0 Time sample 10 20 0 −200 −100 1.4 Impulse Response 0.3.6 0.2 1 0.1 −10 −5 0 Time sample FIR Filter L=20 0.8 0.4 Impulse Response 0.3.5) we analyze the effect of tapering the impulse response of the ﬁlter before computing the amplitude response.2 0.2 5 10 0 −200 −100 1.6 0.3 0. It is clear that the oscillations around the transition band have been eliminated. ON THE DESIGN OF FINITE IMPULSE RESPONSE FILTERS 75 FIR Filter L=10 0.4 1.8 0.3 0.4 1. therefore ﬁlters that are too short might not quite reﬂect the characteristics of the desired amplitude response. The ﬁlter were obtained by truncating the ideal inﬁnite length impulse response sequence. In ﬁgure (3.2 0.4: Impulse response of two ﬁnite length ﬁlters and the associated amplitude response.5 Amplitude Response 0. 4 ¡ B § P) ¥¥ B F4 ¡ ¥ GE ¡ 1 ¡ ¦ $ © " ' E © " ' § E B 3 " © .2 FIR Filter L=10 0 f(Hz) FIR Filter L=20 100 200 0 f(Hz) 100 200 Figure 3.

6 0.9 0.2 0.1 0 −20 −15 −10 −5 0 Sample 5 10 15 20 B ¦ ¨ 1 Figure 3.3 0.4 0.1 0 −0. In this case the truncated impulse response was taper with a Hamming window.4 1.1 −20 −10 0 Time sample 10 20 0 −200 −100 1.7 0.4 Impulse Response 0.1 0 −0.8 0.6 0.3 0.6: A Hamming taper (window) of length .2 5 10 0 −200 −100 1.3 0.2 1 0.4 0. Tapering helps to attenuate side-lobe artifacts (Gibbs phenomenon) Hamming window L=20 1 0.2 0. DISCRETE FOURIER TRANSFORM FIR Filter L=10 0.2 1 0.5 0.8 0.5 Amplitude Response 0. The ﬁlter were obtained by truncating the ideal inﬁnite length impulse response sequence.5: Impulse response of two ﬁnite length ﬁlters and the associated amplitude response.4 1.4 0.4 Impulse Response 0.2 0.8 0. .76 CHAPTER 3.2 FIR Filter L=10 0 f(Hz) FIR Filter L=20 100 200 0 f(Hz) 100 200 Figure 3.5 Amplitude Response 0.1 −10 −5 0 Time sample FIR Filter L=20 0.6 0.

3. If the amplitude response of a low pass ﬁlter if given by we can contrstruct a high pass ﬁlter with the same cut-off frequency using the following expression: that suggests that one can compute the impulse response of the low pass ﬁlter an then transform it into a high pass ﬁlter using the following expression: ¥ 2 ¡0 ¡ ' ' § §§ 7 7 ¥ 210¡ E B 32 ¡0 §¥ 5 E B § 5 5 E § 5 (3.2 High Pass ﬁlters Knowing how to compute low pass ﬁlters allows us to compute high pass ﬁlters.44) .3.43) (3.3. ON THE DESIGN OF FINITE IMPULSE RESPONSE FILTERS 77 3.

DISCRETE FOURIER TRANSFORM .78 CHAPTER 3.

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