Lecture 9

Discrete Time
Fourier
Transform
(DTFT)

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 Today in DSP
 The Discrete Time Fourier Transform
ª DTFT properties

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Properties of DTFT
 We will study the following properties of the DTFT:
ª Linearity Î DTFT is a linear operator
ª Time reversal Î x[n] ÅÆ X(-ω)
ª Time shift Î x[n-n0] ÅÆ X(ω)e-jωn 0

ª Frequency shift Î x[n] ejω nÅÆ X(ω- ω0)

0

ª Convolution in time Î x[n]*y[n] ÅÆ X(ω).Y(ω)

ª Convolution in frequency
ª Differentiation in frequency Î nx[n] ÅÆ j (dX(ω)/dω)
ª Parseval’s theorem Î Conservation of energy in time and frequency domains

ℑ ℑ
x[n] ⇔ y[n] ⇔ Y (ω
X (ω ) )
Digital Signal Processing, © 2006 Robi Polikar, Rowan University
 Linearity &
Differentiation In Frequency
 The DTFT is a linear operator

ℑ
ax[n] + by[n] ⇔ aX (ω ) + bY
(ω )
 Multiplying the time domain signal with the independent time variable
is equivalent to differentiation in frequency domain.

ℑ
dX
nx[n] ⇔ j
(ω
dω)

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Time Reversal ,
Time & Frequency Shift
 A reversal in of the time domain variable causes a reversal of the
frequency variable ℑ
x[−n] ⇔ X (−ω
)
 A shift in time domain by m samples causes a phase shift of e-jωm in
the frequency domain
ℑ
x[n − m] ⇔ X (ω )e − jωm
ª Note that the magnitude spectrum is unchanged by time shift. Why?

 A shift in frequency domain by ω0 causes a time delay of ejω n

0

ℑ
n
x[n]e jω 0 ⇔ X (ω − ω0 )

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Convolution
 Convolution in time domain is equivalent to multiplication in
frequency domain
ℑ
x[n]* h[n] ⇔ X (ω ) ⋅ H
(ω )
ª This is one of the fundamental theorems in filtering. It allows us to compute the
filter response in frequency domain using the frequency response of the filter.

 Multiplication in time domain is equivalent to convolution in

frequency domain
ℑ 1 π
x[n] ⋅ h[n] π ∫ X (γ )H (ω −
⇔ 2 −π γ )dγ

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Parseval’s Theorem
 The energy of the signal , whether computed in time domain or the
frequency domain, is the same!

∞ 1 π 2 2
∑ x[n] = 2π ∫X dω
n = −∞ −π(ω )

Energy of a continuous
periodic function

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Important DTFT Pairs
Impulse Function
 The DTFT of the impulse function is “1” over the entire frequency
band.
ℑ{δ [n]} = 1

Extend of the frequency band

in discrete frequency domain

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Important DTFT Pairs
Constant Function
 Note that x[n]=1 (or any other constant) does not satisfy absolute
summability. However, we can sow that the DTFT of the constant
function is an impulse at ω=0. (this should make sense!!!)
∞
ℑ{1} 2π ∑ δ (ω −
m=
= 2πm )
−∞

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Matlab Approximation
 In class demo!

x=zeros(1000,1); x=ones(1000,1);
x(500)=1; subplot(211)
subplot(211) plot(x); grid
plot(x); grid X=abs(fft(x));
X=abs(fft(x)); subplot(212)
subplot(212) w=linspace(-
w=-pi:2*pi/999:pi; pi, pi, 1000);
plot(w, fftshift(X)); grid plot(w, fftshift(X)); grid

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Important DTFT Pairs
The Complex Exponential
 The DTFT of the complex exponential:
∞

x[n] = e jω n ⇔ X (ω = 2π ∑ δ (ω − ω0 + 2π k )
0

)
k =−∞

We are only interested in

[- π π] range, where there is
only one spectral component
Hence, the spectrum of a single
complex exponential at a specific
frequency is an impulse at that ... ...
frequency.
ω0-4π ω0-2π ω0 ω0+2π ω0+4π
This can be verified by computing
the inverse DTFT of X(ω) given
above.

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Important DTFT Pairs
Real Exponential
 This is an important function in signal processing. Why?
ℑ 1
n
x[n] = α u[n] ⇔
1 − α e − jω
t=0:0.01:10;
x=(0.5).^t;
plot(t,x)
X=fftshift((fft(x)));
subplot(311)
plot(t,x); grid
subplot(312)
plot(abs(X)); grid
f=-50:100/1000:50;
plot(f,abs(X)); grid
subplot(313)
plot(f, unwrap(angle(X))); grid

In Matlab, periodicity of X(ω) is assumed

Digital Signal Processing, © 2006 Robi Polikar, Rowan University
 Important DTFT Pairs
The sinusoid at ω0
 By far the most often used DTFT pair (it is less complicated then it looks):
ℑ ∞ ∞
x[n] = cos(ω0 n ) ⇔ π ∑ δ (ω− 2πm − ω0 ) + π ∑ δ (ω− 2πm + ω0 )
m= −∞ m= −∞

ℑ ∞ The above expression can also be obtained

x[n] = e jω0 n
⇔ 2π ∑ δ ( ω − ω0 ± 2 π m ) from the DTFT of the complex exponential
m=−∞ through the Euler’s formula.
Digital Signal Processing, © 2006 Robi Polikar, Rowan University
 Important DTFT Pairs
Rectangular Pulse
 Also very commonly used in DSP, as it provides the FT of an ideal
lowpass filter (we will see this later)

⎧1, − M ≤ n ≤
x[n] = rect M [n] = ⎨
M
⎩0, otherwise
ℑ
M
⇔
∑ =sin (M , ω≠0
e − jω n sin ( 2 )
+ 1 2 )ω ω
n=−M

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Ideal Lowpass Filter
 The ideal lowpass filter is defined as

⎧1, ω ≤ω
H (ω ) ⎨
c
⎩ 0, ωc ≤ ω ≤
=
π
ª Taking its inverse DTFT, we can obtain the corresponding impulse function h[n]:

sin ωc n
h[n] =
πn

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Ideal Lowpass Filter
 Note that:
ª The impulse response of an ideal LPF is infinitely long Æ This is an IIR filter.
In fact h[n] is not absolutely summable Æ its DTFT cannot be computed Æ an
ideal h[n] cannot be realized!
ª One possible solution is to truncate h[n], say with a window function, and then
take its DTFT to obtain the frequency response of a realizable FIR filter.

Digital Signal Processing, © 2006 Robi Polikar, Rowan University

 Some Useful
Matlab Functions
 Matlab cannot explicitly calculate the DTFT, since the frequency axis is continuous.
However, it can calculate an approximation of the DTFT using a given number of points.
 y=fft(x, N) – Calculates the discrete Fourier transform of the signal x at N points. If N is not
provided, length of y is the same as x. DFT is a sampled version of the DTFT, where the
samples are taken at N equidistant points around the unit circle from 0 to π
 [h,w] = freqz(b,a,N,'whole') – Calculates the frequency response of a filter whose CCLDE
coefficients are given as b and a, using N number of points around the unit circle. If ‘whole’
is included, it returns a frequency base of w from 0 to 2π, otherwise, from 0 to π.
 y=abs(x)- Calculates the absolute value of signal x. For complex values signals, the output is
the magnitude (spectrum) of the complex argument.
 y=angle(x) – Calculates the phase (spectrum) of the signal x.
 q = unwrap(p) corrects the radian phase angles in a vector p by adding multiples of 2π
when absolute jumps between consecutive elements of p are greater than the default jump
tolerance of π radians.
 y = fftshift(x) rearranges the outputs of fft by moving the zero-frequency component to the
center of the array. It is useful for visualizing a Fourier transform with the zero-frequency
component in the middle of the spectrum.

Digital Signal Processing, © 2006 Robi Polikar, Rowan University