5.

9 DSP II:
Math Background
Dr. Tarek A. Tutunji
Mechatronics Engineering Department
Philadelphia University, Jordan
 Overview
 In introduction to DSP was given in the
previous sequence

 In this sequence, basic mathematical

background of common used algorithms in
DSP is provided. This will include
• Correlation
• Convolution
• Digital Filtering
• Power Spectrum
 Classification of
Discrete-Time systems
 Static vs. Dynamic

 Time-invariant vs. time-varying

 Linear vs. nonlinear

 Causal vs. noncausal

 Stable vs. nonstable

We are interested in LTI (Linear Time-Invariant) causal Systems

 Correlation
 Correlation between two signals is a measure of the
degree to which the two signals are similar and for how
long they remain similar when one is shifted with respect
to the other.
• Correlation is maximum when the signals are similar in
shape and are in phase

 The cross-correlation of two signals, x(n) and y(n), is

given by

r xy (k)   x(n)y(n  k)
k - 

• One signal is shifted with respect to the other

• Each element of one signal is multiplied by the corresponding
element of the other
• The area under the resulting curve is integrated
• Correlation requires a lot of calculations.
 Convolution
 Consider a system h(n) with input x(n) and output y(n)

x(n) h(n) y(n)

Then, y(n)  x(n)* h(n)

 If the input is a pulse d(n)  the output y(n) = h(n)

 Convolving two signals is equivalent to multiplying the

frequency spectra of the two signals together

y(n)  x(n) * h(n)

Y(Z)  X(Z)H(z)
 Convolution
 Convolution is used for digital filtering.

 The convolution of two signals, x(n) and h(n), is given by


y(n)  x(n)* h(n)   x(n)h(n  k)
k - 

• Folding, h(k) around 0 to get h(-k)

• Shifting, h(-k) by n0 to right (left) if n0 is positive (negative)
• Multiply x(k) by h(n0-k)
• Sum all values at time n=n0
 Fourier Transform
 Most signals can be decomposed into sum of sinusoidal signal
components (with the appropriate amplitude, frequency, and
phase)
• For class of periodic signals, the decomposition is called Fourier Series
• For the class of finite energy signals, the decomposition is called the
Fourier Transform

 The Fourier transform is an equation to calculate the frequency,

amplitude and phase of each sine wave needed to make up any
given signal.

 The Fourier Transform (FT) is a mathematical formula using

integrals

 The Discrete Fourier Transform (DFT) is a discrete numerical

equivalent using sums instead of integrals

 The Fast Fourier Transform (FFT) is a computationally fast way to

calculate the DFT
 Fourier Transform
 Z-Transform (one-sided / causal) is


X(z)   x(n)Z - n
n 0

 By letting z = ejw and w = 2pf = 2pk/N


X(e jω
)  x(n)e - jω
n 0
 2π 
N 1 - j  kn
Xk   x(n)e  N 
Where Xk is the Discrete
n 0 Fourier Transform (DFT)
 Digital Filters
 There are two main kinds of filter, analog and digital.
 An analog filter uses analog electronic circuits as explained
in session 3 (signal conditioning)
 A digital filter uses a digital processor to perform numerical
calculations on sampled values of the signal.
 The analog input signal must first be sampled and digitized
using an ADC
 In a digital filter, the signal is represented by a sequence of
numbers, rather than a voltage or current.
 The following diagram shows the basic setup of such a
system.
 Analog vs. Digital Filters
Analog Digital
Noise Immunity No Yes
Programmable No Yes
Design Difficult Easy
# of Components Large Small
# of Applications Small Large
Resolution Low High
Modification Difficult Easy
Sensitivity High Low
High Frequency Yes Yes
Low Frequency No Yes
 ARMA model
 Auto Regressive Moving Average (ARMA) model with
input x(n) and output y(n) is given below
n m
y(n)    ak y(n  k)   bk x(n  k)
k 1 k 0

 By taking the z-transform and manipulating the result, we get

n m
Y(Z)    ak Z Y(z)   bk Z - k X(z)
-k
k 1 k 0
n m
 Y(z)   ak Z - k Y(z)   bk Z - k X(z)
k 1 k 0
n m
 Y(z)(1   ak Z -k
)  X(z)(  bk Z - k )
k 1 k 0
m
 bk Z - k
Y(z) k 0
  H(z) 
n
which is the Transfer Function
X(z)
1   ak Z - k
k 1
 IIR and FIR Filters
 Infinite Impulse Response (IIR) filters contain
both poles and zeros and therefore depend on
inputs and delayed outputs. So in the time
domain the are ARMA

 Finite Impulse Response (FIR) filters contain only

zeros (i.e. no poles) and therefore depend only
on inputs. So in the time domain they have only
the MA (Moving Average) part

 Designing digital filters is the process of finding

the appropriate coefficients (a’s and b’s) to
obtain the desired system response
 Example
 Consider the ARMA equation
y(n)  x(n)  0.7y(n  1)

x(n) y(n)
+
0.7
Z-1

 The transfer function is

1
H(z) 
1  0.7z 1
Example: Pole-Zero Plot

Pole at 0.7
Example: Frequency Response
Low pass filter
Example: Impulse Response

Stable system since

Pole < 1
 Power Spectrum
 Spectral Analysis
• Captures the frequencies present in a signal
• Estimates the sine waves that can be added to create a
duplicate of a given signal

 Power Spectrum (or Frequency Spectrum)

• Decomposes a signal into its basic frequency components
• Calculates the power in each of those frequencies
• Shows the distribution of the power in the frequency domain

 Understanding the relation between time and frequency

domains is useful:
• some signals are easier to visualize in the frequency domain
while others are easier to visualize in the time domain
• Some signals take less information to define in the time
domain while others take less information to define in the
frequency domain
 Power Spectrum
 The Power Density Spectrum is the Fourier
Transform of the autocorrelation function
N 1
 j2pfm
Pxx (f)   xx
r (m)e
m (  N 1)

 We can also calculate the power spectrum from

the DFT as follows

1 2
Pxx ( k )  Xk
N
 Example
 Consider the time domain signal
 Example
 The Power Spectrum is given below
3rd component at k=13
4th component at k=18 mirror image
2nd component
at k=5

1st component
at k=2

 The original frequencies in the signal depend on Sampling

used, frequency = (k/50)*Fs
 So if Fs = 4KHz  1st frequency is 160 Hz
 Summary
 The basic theory and math behind
commonly used DSP algorithms were
provided. This included
• Correlation
• Convolution
• Digital Filters
• Power Spectrum