DSP Ai

Air University Islamabad Department of Telecommunication Engineering. Prof. Dr.
Kamal Athar
Digital Signal Processing

course lecture notes
Lecturer Asim Ali Samejo
September 2005
II
Contents
1 Introduction 1.1 1.2 Signals,systems and signal processing . . . . . . . . . . . . . . . . . . . . . DSP applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1 1 3
2 Classication of systems and signals 2.1 Classication of signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2.1.1 2.1.2 2.1.3 2.2 2.3 Continuous-time signals . . . . . . . . . . . . . . . . . . . . . . . . Discrete-time signals . . . . . . . . . . . . . . . . . . . . . . . . . . Common test signals . . . . . . . . . . . . . . . . . . . . . . . . . .
6 6 7 7 7 9
Classication of systems . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Digital processing of continuous signals . . . . . . . . . . . . . . . . . . . . 11 2.3.1 2.3.2 2.3.3 Sampling: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14 Quantization: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16 Characteristics of A/D converter . . . . . . . . . . . . . . . . . . . 24
III
3 Transform domains 3.1
25
Fourier transforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25 3.1.1 3.1.2 3.1.3 3.1.4 3.1.5 Denitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25 Existence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26 Periodicity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26 Characteristics, Rules, Theorems . . . . . . . . . . . . . . . . . . . 27 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
3.2
Z transforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 3.2.1 3.2.2 3.2.3 3.2.4 3.2.5 Denitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 Existence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34 Periodities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37 Inverse Z-transform . . . . . . . . . . . . . . . . . . . . . . . . . . . 38 Characteristics rules,theorems . . . . . . . . . . . . . . . . . . . . . 39
3.3
Discrete Fourier transforms . . . . . . . . . . . . . . . . . . . . . . . . . . 41 3.3.1 3.3.2 3.3.3 3.3.4 3.3.5 3.3.6 3.3.7 Denitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41 Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42 Linear and circular convolution . . . . . . . . . . . . . . . . . . . . 44 Frequency analysis of stationary signals . . . . . . . . . . . . . . . . 46 Ecient computation of DFT, FFT . . . . . . . . . . . . . . . . . . 54 Decimation-in-time FFT algorithm . . . . . . . . . . . . . . . . . . 54 Transformation of real-valued sequences . . . . . . . . . . . . . . . 61 IV
4 Digital lters 4.1
64
Digital lter structures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64 4.1.1 4.1.2 Structures of FIR lters . . . . . . . . . . . . . . . . . . . . . . . . 66 Structures of IIR lter . . . . . . . . . . . . . . . . . . . . . . . . . 69
4.2
Coecient quantization and roundo eects . . . . . . . . . . . . . . . . . 76 4.2.1 4.2.2 Coecient quantization . . . . . . . . . . . . . . . . . . . . . . . . 81 Round-o eects . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
4.3 4.4
Computation of frequency response . . . . . . . . . . . . . . . . . . . . . . 91 Characteristics of practical frequency selective lters 4.4.1 4.4.2 . . . . . . . . . . . . 93
Design of FIR lters . . . . . . . . . . . . . . . . . . . . . . . . . . 95 Design of IIR lters . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
4.5
Analog to digital transformation methods . . . . . . . . . . . . . . . . . . . 100
5 Multirate signal processing 5.1 5.2 5.3
106
Sampling rate reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106 Sampling rate enhancement . . . . . . . . . . . . . . . . . . . . . . . . . . 106 Filter banks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
Literaturverzeichnis
107
VI
Chapter 1 Introduction
1.1
Signals,systems and signal processing
What does Digital signal processing mean? Signal:
Physical quantity that varies with time, space or any other independent variable. Mathematically: Function of one or more independent variables s1 (t) = 5, s2 (t) = 20t2 Examples: Temperature over time t, brightness of image over (x, y ) pressure of sound over (x, y, z ) or (x, y, z, t).
Signal Processing:
Passing the signal through a system. Examples:
CHAPTER 1. INTRODUCTION - Modication of the signal (ltering, noise reduction, equalization,. . . ) - Prediction,transformation to another domain(e.g. Fourier transform) - Determination of signal statistics mean value, variance and p.d.f.,. . .
Properties of the system (e.g. linear/nonlinear) determine the properties of whole processing operation.
System: Denition also includes:
- software realizations of operations on a signal, which are carried out on a digital computer ( software implementation of system) - digital hardware realizations (logical circuits) congured such that they are able to perform the processing operations, or - most general denition: a combination of both Digital signal processing: Processing of signals by digital means (software and/or hardware) Includes Conversion from analog to digital domain and back (physical signals are analog) Mathematical specication of the processing operations Algorithm: method or set of rules for implementing the system by a program that performs the corresponding mathematical operations. Emphasis on computationally ecient algorithms, which are fast and easily implemented. why Digital?
CHAPTER 1. INTRODUCTION
Property Precision
Digital generally unlimited cost, complexity precision
Analog generally limited increase in performance drastic increase in cost Problematic Higher approximate realization
Aging production costs Linear-phase frequency response Complex Algorithms
Without problems Low exactly realizable
realizable
strong limitations
1.2
DSP applications
- Military applications: Targeting, sonar, radar, secure communications. - Telecommunications: Echo cancellation, speech coding, modems, channel estimation. - PC and Multimedia applications: audio, video on demand, voice recognition and synthesis. - Entertainment: Audio, video compression e.g.(mp3, mpeg, dvd, cd player) - Automotives: active background cancellation, navigation.
Historical perspective
- Sampling theorem(nyquist) 1920s - PCM was established in 1940s - Digital ltering, FFT, speech analysis 1960s (MIT, IBM, BELL Labs) - Adaptive lters and linear prediction 1960s (stanford) - Digital spectral estimation and speech coding 1970s
CHAPTER 1. INTRODUCTION - First generation DSP chips from Intel,TI,AT&T, motorola and analog devices
- Low cost DSP late 1980s - high speed and complexity, media applications, low power and portable today.
CHAPTER 1. INTRODUCTION
Examples
Chapter 2 Classication of systems and signals
2.1
Classication of signals
Signal can be classied into periodic and aperiodic functions in time. Another classication refers to continuity or discreteness of time variable. A continuous-time signal has a value f (t) associated with every instant in time. In contrast Discrete-time signal only exists at discrete points t = kT of real time, where T is interval between adjacent sampling points.
C/D converter
p(t )
Conversion from impluse train to discrete sequence
va (t )
vi (t )
v(k)
va (t )
T = T1
T = 2 T1
3T 2T T
T 2T 3T
t
v(k)
2T
v(k)
3 2 1 0 1
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS
2.1.1
Continuous-time signals
Continuous time signals are dened for every instance in time. A continuous time signal with continuous amplitude is called analog signal. Example Electrocardiogram(ECG), speech signal, musical signal.
2.1.2
Discrete-time signals
Discrete time signal is called sample. Discrete-time signals are generated from parent continuous signals via sampling. If discrete instants of time at which a discrete-time signal is dened are uniformly spaced, discrete samples can be quantized. If v (t) can be determined by t (possibly uniquely) and if relation can be evaluated despite of a non-random character then v (t) is deterministic. If v (t) is random signal or if relation of v (.) and t can not be evaluated despite a nonrandom character then v (t) is / treated as a stochastic signal.
2.1.3
Common test signals
Unit impulse Unit sample is denoted by 0 (k ) 0 (k ) = 1, k = 0 0, elsewhere (2.1)
0.8
(k)
0.6
0.4
0.2
0 10
0 k
10
It is the elementry digital signal, scaled/delayed impulses can represent any arbitrary digital sequence. x(n) =
k = +
x(k )0 [k ]
(2.2)
Delay/shift and Sifting properties must be considered. Step Unit step sequence is denoted by 1 (k ) 1, k 0 0, k < 0
1 (k ) =
(2.3)
0.8
1(k)
0.6
0.4
0.2
0 20
10
10
20
30
40
50
extension of unit impulse i.e.

+
1 (k ) =
n=0
0 [k n]
(2.4)
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Rectangular pulses
M (k ) =
1, 0 k M 0, elsewhere
(2.5)
0.8
(k)
0.6
0.4
0.2
10
15
20
25 k
30
35
40
45
50
2.2
Classication of systems
system is an operator excited by input sequences, creating internal(state)sequences and output sequences.
Memory
A system is dened to be memoryless if the output y [k ] at every value of k depends only on the input x[k ]. Example: y [k ] = [x(k )]2 AWGN channel, multi-tap channel, capacitor
10
Linearity
A system is said to be linear if the output y [k ] for superposition of two input signals v1 (k ) and v2 (k ) is also a linear superposition of corresponding individual outputs y1 (k ) and y2 (k ). i.e. T {v1 (k ) + v2 (k )} = T {v1 (k )} + T {v2 (k )} = y1 (k ) + y2 (k ) (2.6)
T {av (k )} = aT {v (k )} = aT {y (k )}
(2.7)
Where a is arbitrary constant, equation (2.6) shows additive superposition and (2.7) shows scaling property. Equations can be generalized
T {av1 [k ] + bv2 [k ]} = aT {v1 [k ]} + bT {v2 [k ]}
(2.8)
v1 (k)
S
x
y1 (k)
+
x
ys (k)
v2 (k)
y2 (k)
Obviously y3 (k ) 0 k k if 1 s{v1 (k )} + 2 s{v2 (k )} = s{1 v1 (k ) + 2 v2 (k )} Example:
y (t) = a v 2 (t) + b v (t) + c Remarks: Linear systems are of specially important because of simplicity in mathematical modeling.
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Most real world signal are not linear, on contrary they are non-linear.
11
For linear systems, however, a widely general theory is applicable- therefore its is often tried to deal with actually non-linear system after lineariztion.
causality
A system is causal if only for any instance in time k0 the output y [k0 ] depends only on the input sample k0 and prior. Example: a device which adds up incomming sample values accumulator
k0
y (k 0 ) =
n=
v (n)
(2.9)
stability
A system is said to be stable in bounded-input bounded-output(BIBO) sense if and only if for every bounded input sequence the system produces a bounded output sequence. Input v (k ) is bounded if there exist a nite positive value M1,2 , such that |v (k )| M1 < k (2.10)
then the system is stable if for every bounded-input there exists a positive nite value. |y (k )| M2 < k (2.11)
2.3
Digital processing of continuous signals
Generation of discrete-time signals from the continuous-time signals
p(t) =
k =
0 (t kT )
(2.12)
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Where 0 (t) is dirac(delta)function and T is Sampling Interval
12
vi (t) = va (t)
k =
0 (t kT ) va (kT )0 (t kT ) (2.13)
vi (t) =
k =
How does the Fourier transform F {vi (t)} Vi (j ) look like? fourier tranform of impulse train: 2 p(t) P (j ) = 0 ( ks ) T k= where s = 2/T sampling frequency in radians/sec. Vi (j ) = 1 Va (j ) P (j ) 2 (2.15)
(2.14)
we nally have the Fourier transform of vi (t) 1 Vi (j ) = T
Va (( ks ))
k =
(2.16)
Periodically repeated copies of va (j ), shifted by integer multiplies of the sampling frequency, scaled with factor of
1 . T

Va ( j)
13
n
2 T
P( j)
s
1 T
2s
Vs ( j)
n s
Vs ( j)
1 T
1. Fourier transform of a bandlimited analog input signal Va (j ), highest frequency is N . 2. Fourier transform of dirac impulse train. 3. Result of convolution P (j ) Va (j ). It is evident that when s N N or s > 2N (2.17)
xa (t) can be recovered via ideal low-pass lter. if (2.17) does not hold i.e. if s < 2N copies of Va (j ) would overlap and signal xa (t) cannot be recovered by lowpass ltering.Aliasing. Ideal sampling is practically impossible. 0 (t) can be generated only approximately.
discrete samples x(k ) do not contain any information about sampling interval. Sampling frequency must be known apriori for reconstruction of original signal.

p(t ) = o (t kT )
k
14
va (t )
vi (t )
H ( j )
vr (t )
Va ( j)
n
1 T
Vi ( j) N < s / 2
n s
H ( j )
Vr ( j)
n n
2.3.1
Sampling:
Reconstruction of ideally sampled signal by ideal lowpass ltering: In order to get the input signal va (t) after reconstruction lter, i.e. Vr (j ) = Va (j ) N < s 2 and N < c < (s N ) (2.18)
must be satised!.Then we have Vr (j ) = Va (j ) = Vi (j ) Hr (j ) va (t) = vr (t) = vi (t) hr (t) the frequency response of ideal lowpass lter T, | | < c 0, elsewhere (2.19)
Hr (j ) =
rect(
) hr (t) = sinc(s t/2) c
(2.20)
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Combining (2.13),(2.20) in (2.19) yields

+ +
15
va (t) = =
: k = +
va (kT )0 ( kT )sinc
+
1 s (t ) d 2 1 s (t ) d 2
va (kT )
k = +
0 ( kT )sinc
=
k =
va (kT )sinc
1 s (t kT ) 2
p(t ) = + k= 0 (t kT )
Hr (e j ) vi (t ) vr (t )
va (t )
Va ( j)
1 T
Hr ( j)
n Vi ( j)
c Vr ( j)
1
1 T
sampling theorem: Every bandlimited continuous-time signal va (t) with N < s /2 can be uniquely recovered from its samples through:
va (t) =
k =
va (kT )sinc
1 s (t kT ) 2
(2.21)
16
equation (2.21) is called the ideal interpolation formula, and sinc is termed as ideal interpolation function. reconstruction of continuous signal using ideal interpolation.
va (t )
t
vi (t )
t
vr (t )
2.3.2
Quantization:
Conversion carried out by an A/D-converter involves quantization of the sampled input signal v (kT ) = vi (kT ) and the encoding of result into binary representation. Quantization is a nonlinear and noninvertible process which realizes the mapping.
v (kT ) = v (k ) vn where vn is taken from a nite alphabet I .
(2.22)
Signal amplitude range is divided into L intervals In , using L + 1 decision levels d1 , d2 , . . . , dL+1 In = {dn < v (k ) dn+1 }, n = 1, 2, . . . , L
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Mapping in (2.3.2) is denoted as vq (k ) = Q[v (k )].
I3
Quantization level Decision level
17
instantenous amplitude Range of quantizer
Uniform or linear quantizers with constant quantization step size are very often used in signal processing applications.
= vn+1 vn = const, n = 1, 2, . . . , L 1 Midtreat quantizer: Zero is assigned a quantization level Midrise quantizer: Zero is assigned a decision level.
Q[v]
Twos Complement binary offset
011 010
111 110 101 100 011 010 001 000
2b+1
2 9 2 7 2 5 2 3 2 2 3 2 5 2 7 2
001
000 111 110 101 100
Transfer characteristic for midtreat quantizer
Quantization error eq (k ) with respect to the unquantized signal < eq (k ) 2 2
(2.23)
18
If the dynamic range of the input signal (vmax vmin ) is larger than the range of quantizer, the samples exceeding the quantizer range are clipped, which leads to eq (k ) >
. 2
Quantization characteristic function for a midtreat quantizer with L = 8.
Coding
The coding process in an A/D converter assigns a binary number to each quantization level. with wordlength of b bits we can represent 2b > L binary number to each quantization level.
b log2 (L) the step size or the resolution of the A/D converter is given as = of the quantizer. Twos complement representation is used in most xed-point DSPs: A b bit binary fraction [0 1 2 . . . b1 ],0 denoting the most signicant bit (MSB) and b1 the least signicant bit (LSB), has value
b1 R 2b
with range R
v = 0 +
l=1
l 2l
(2.24)
number representation has no inuence on the performance of quantization process.
Quantization errors
Quantization error is modeled as noise, which is added to the unquantized signal
19
number +7 +6 +5 +4 +3 +2 +1 0 0 1 2 3 4 5 6 7 8
+ve ref +7 8 +6 8 +5 8 +4 8 +3 8 +2 8 +1 8 0+ 0 1 8 2 8 3 8 4 8 5 8 6 8 7 8 8 8
-ve ref 7 8 6 8 5 8 4 8 3 8 2 8 1 8 0 0+ +1 8 +2 8 +3 8 +4 8 +5 8 +6 8 +7 8 +8 8
sign magn 0111 0110 0101 0100 0011 0010 0001 0000 1000 1001 1010 1011 1100 1101 1110 1111
2s compl. 0111 0110 0101 0100 0011 0010 0001 0000 (0000) 1111 1110 1101 1100 1011 1010 1001 (1000)
oset bin. 1111 1110 1101 1100 1011 1010 1001 1000 (1000) 0111 0110 0101 0100 0011 0010 0001 (0000)
1s compl. 0111 0110 0101 0100 0011 0010 0001 0000 1111 1110 1101 1100 1011 1010 1001 1000
20
Quantizer
Q(v)
v +
v+
Modeling of quantization noise
Assumptions: < eq (k ) The quantization error eq (k ) is uniformly distributed over the range 2 The error sequence eq (k ) is modeled as a stationary white noise sequence. The error sequence eq (k ) is uncorrelated with the signal sequence v (k ). The signal sequence is assumed to have zero mean. Assumptions do not hold in general, but they are fairly well satised for large quantizer word-lengths b. Eect of quantization error or quantization noise on the resulting signal vq (k ) can be evaluated in terms of signal-to-noise ratio (SNR) in decibels(dB). SNR = 10log10 Pv Pn (2.25)
. 2
Where Pv denotes the signal power and Pn denotes the power of the quantization noise. In case of quantization with rounding, noise is assumed to be uniformly distributed in the range(/2, /2).
pr (e)
1
= 2 b
Er
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Zero mean, and a variance of

2 = Pn = e /2 /2
21
e2 p(e)de =
/2 /2
e2 de =
2 12
(2.26)
Inserting the denition of in (2.26) yields,

2 = e
22b R2 12
(2.27)
using (2.25) we obtain SNR = 10 log10

2 2 v 12 22b v = 10 log 10 2 e R2 R dB = 6.02b + 10.8 20 log10 v
(2.28)
Explanation of (2.28)
v root-mean-square(RMS) amplitude of the signal v (t). v small decreasing SNR. furthermore not evident from (2.28) v large range R is exceeded. Signal amplitude has to be carefully matched to the range of the A/D converter. For music and speech a good choice is v = R/4. Then the SNR of a bbit quantizer can be approximately determined as SNR = 6.02b 1.25dB Each additional bit in the quantizer increases the signal-to-noise ratio by 6 dB. Examples: Narrowband speech: b=8 Bit SNR=46.9 dB Music(CD): b=16 Bit SNR=95.1 dB Music(Studio): b=24 Bit SNR=143.2 dB (2.29)
22
A/D converters
Flash A/D converter
Vin +VRe f
R
V6
R
V5
R
V4
R
83 Priority Encoder
MSB
V3
R
LSB
V2
R
V1
R
V0
R
VRe f
start
Vin
Control
logic
Clock
Nbit shift register
N
Nbit shift register
Nbit
D/A Converter
VD
VRe f
Analog input voltage VA is simultaneously compared with a set of 2b 1 separated references voltages by means of analog comparators locations of the comparators circuits indicate the range of the input voltage.
CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS All output bits are developed simultaneously very fast conversion. (1-1.5GHz) Hardware requirements increase exponentially with linear increase in resolution.
23
Flash converters are used for low-resolution(b < 8bits) and high speed conversion applications.
Serial to parallel A/D converters

Here, two b/2bit ash converters in a serial-parallel conguration are used to reduce the hardware complexity at the expense of slightly higher conversion time.
2 N /2
Vin
VRe f
N/2bit ADC
N/2bit DAC N/2 bit
VRe f VRe f
N/2bit ADC N/2 bit LSB
Subranging ADC
MSB
Vin
VRe f
N/2bit ADC
N/2bit DAC N/2 bit

VRe f
VRe f
N/2bit ADC N/2 bit LSB
MSB Ripple AD converter
Vin
Sample and Hold
+
MSB Flash Encoder
DAC
LSB Flash Encoder
Digital error
Corrector
(Adder)
Digital output
24
2.3.3
Characteristics of A/D converter
111 110
Digital code
111 110
2
b+1
100 011 010 001 000 analog input

2b+1 2
Digital code
101
101 100 011 010 001 000

2b+1 2
2b+1
analog input
Chapter 3 Transform domains
Introduction
Spectral representation means generally: representation of a sequence by a linear superpositions of suitable basis sequences, weighted by suitable factors.
Spectral analysis means generally: determination of above named weighting factors for a given sequence and basis. Spectral synthesis means generally the above-named linear superposition. This is very simple yet very useful representation of deterministic and stochastic signals.
3.1
3.1.1
Fourier transforms
Denitions
v (k ), k Z, v C V (ej ) =
k =
v (k )ej k
(3.1)
25
CHAPTER 3. TRANSFORM DOMAINS
26
is the Spectrum of v (k ). Equation (3.1) is called the Discrete-time Fourier transform of v (k ). Inverse transformation 1 V (e ), R , V C v (k ) = 2
j
V (ej )e+j k d

(3.2)
Equation (3.2) = spectral synthesis i.e. v (k ) written as superposition of harmonics of ej k weighted by

1 2
V (ej )d as Complex amplitudes V (ej )= amplitude density
3.1.2
Existence
3.1 = nite summation existence |V (ej )| M <

+ k = + +
| V (e )| = |
j
3.1
v (k )e
j k
|
k =
| v (k )| | e
j k
|=
k =
| v (k )|
(3.3)
so: If
+ k =
|v (k )| M < , i.e.
v (k ) is absolutely
summable then
|V ej | M < holds certainly. But: This is only a sucient condition, not a necessary one. Equation (3.3) may be violated while still V (ej ) may exist because
+
| V (e )|
j k =
| v (k )|
3.1.3
Periodicity
in 3.1 = +
k = +
v (k )ejk(+2) = V (ej (+2) ) v (k )ej k ejk(2)

1 V (ej )
=
k =
(3.4)
Fourier transform of a sequence is 2 periodic in .
27
3.1.4
Characteristics, Rules, Theorems
Background
Relation of F transform V0 (j ) = F {v0 (t)} =
+ t=
v0 (t)ejt dt
(3.5)
of a continuous signal v0 (t). If v (k ) contains equi-spaced samples of v0 (t) then, v0 (kT ) = v (k ) V (ej ) =
F
1 T
V 0 j (
=
2 ) T
=T
(3.6)
If we increase T to a large value, it will result in overlapping, known as aliasing.
V ( j)
1 T
4T
3T
2T
2 T
3 T
4 T
a. Linearity: The Fourier transformation is linear in the sense that for v1,2 (k ) V1,2 (ej ) We have and v (k ) = 1 v1 (k ) + 2 v2 (k ) (3.7)
v (k ) V (ej ) = 1 V1 (ej ) + 2 V2 (ej )
The equivalence holds for the inverse transformation obviously (3.7) says F {superposition} = superposition of F s b. Modulation: Given: v (k ) V (ej ) def: v1 (k ) = v (k ) e+j 0 k Complex modulation then from 3.1 we have:
v1 (k ) V [ej (0 ) ]= Shift in frequency
(3.8)
CHAPTER 3. TRANSFORM DOMAINS c. Shift: Given: v (k ) V (ej ) v1 (k ) = v (k k0 )( delayed / advanced / shifted signal )
28
then from 3.1 v1 (k ) V (ej )ej k0 additional linear phase (3.9) The Shift operator will turn out to be a central element of the discrete systems to be dealt with. It does not alter the signal form - and the spectrum neither, except
for adding a linear term (k0 , ) to the phase angle arg {V (ej )} d. Component symmetries: It is well established that v (k ) C v (k ) = {v (k )} + j {v (k )} = vre (k ) + vimg (k )
Any signal can also be decomposed into an even and odd component: 1 v (k ) ve (k ) = [v (k ) + v (k )] = ve (k ) 2 1 vo (k ) = [v (k ) v (k )] = vo (k ) 2 - Both decompositions can be combined. - The same holds for the (continuous) functions V (ej ), over . So: 4 components of v (k ) and V (ej ) each. then from 3.1, unique symmetry relations are found: v (k ) = vRe (k ) + vRo (k ) + jvIe (k ) + jvIo (k ) (3.10) (3.11)
V (ej ) = VRe (ej ) + VRo (ej ) + jVIe (ej ) + jVIo (ej ) (for proof see exercise) v (k ) vIe (k ) 0, vIo(k ) 0 VIe (ej ) 0, VRo (ej ) 0 V (ej ) = VRe (ej ) + jVIo (ej ) = V (ej ) real-valued signals have Hermitian Fourier transforms. Consequences:
3.11
(3.12)
(3.13)
2 j 2 |V (ej )| = VRe (ej ) + VIe (e ) even due to sqr.
= even
(3.14)
The modulus is an even function of if v (k )
CHAPTER 3. TRANSFORM DOMAINS e. Convolution:Linear convolution of continuous function well-known v1 (t) v2 (t) =
=
29
v1 ( ) v2 (t )d = v2 (t) v1 (t)
(3.15)
same denitions for any continuous function for any variable e.g. v1 (j ) and v2 (j ) but for v1,2 (ej ) their periodicities have to be taken into account, so now the denition for periodic function needed, convolution of two periodic spectra.
v1 (e )
v2 (e ) =
v1 (ej ) v2 (ej )d = v2 (ej )
v1 (ej )
(3.16)
circular (periodic) convolution results in 2 periodic sequence. Convolution of continuous spectra can be easily extended to discrete case. Spectral operations on V1,2 (ej ) are equivalent to the convolutions of v1,2 (k ) and viceversa
+ +
F {v (k )} = F {v1 (k ) v2 (k )} =
k = = +
v1 () v2 (k )ej k
+
=
=
v1 ()
k = +
v2 (k )ej (k) ej
+
insert = k V (e ) =
= V1 (ej ) j
v1 () e
v2 ( ) ej = V1 (ej ) V2 (ej )
V2 (ej )
so: v1 (k ) v2 (k ) V1 (ej ) V2 (ej )

F
(3.17)
1 {V1 (ej ) F
V2 (ej )} =
1 2
V1 (ej ) V2 (ej () )mod2 ej k d d
= v1 (k ) v2 (k )
(3.18)
CHAPTER 3. TRANSFORM DOMAINS f. Dierence & Integration given v (k ) v1 (k ) so: V (ej )
30
d dV (ej ) = = ?? V1 (e ) = v (k )ej k = jk v (k ) d d k=
j
jk v (k )
d V (ej ) d
(3.19)
Inversely as there is no derivate of sequence v(k) we consider dierence signal Given: v (k ) v1 (k ) V (ej ) = v (k ) v (k 1) V1 (ej ) =?? (3.20)
V1 (ej ) = V (ej ) 1 ej 1 v (k ) v (k 1) g. parsevals theorem given v1 (k ) V1 (ej ) and v2 (k ) V2 (ej ) then

+ k = v1 (k )v2 (k )
1 2
V1 (ej )V2 (ej )d
(3.21)
proof: inserting (3.1)in(3.21). = 1 2

+ + + k =
v1 (k )ej k V2 (ej )d
(3.22)
=
k = +
v1 (k )v2 (k ) +
| v (k )|
k =
1 2
|V (ej )|2 d
(3.23)
Sum of signal power = Sum of power of components
31
Parseval theorem of (3.23) states that the sum of signal power values gives the same as the sum of component powers.
3.1.5
Examples
1. Impulse (dirac function) (see also 2.1.3) 1, k = 0 0 (k ) = 0, elsewhere

+ k = +
0 (e ) =
3.1
0 (k )e
0 (k )e0 = 1
k =
0 (k ) 0 (ej ) = 1
(3.24)
0 (k) 0 (e j )
0 k
2. Step function:
+ +
1 (k )
k =0
1 (k )e
+
j k
=
k =0
cos(k ) jsin(k ) =???
| 1 (k )|
k =
sucient condition (3.3) is not satised there is no FT (at least common type FT) 1 + (e ) = 1 ej
j
0 ( + 2k )
k =
CHAPTER 3. TRANSFORM DOMAINS 3. Rectangle : M (k ) RM (e ) =

j k =0
32
M 1
1 ej k
Remarks: nite summations no problems with (absolute) summability. Their F - transforms always exist due to (3.3) General summation formula for exponential sequence
k2 k 2 k 1
q k = q k1
k =k 1 k =0
q k = q k1
1 q k2 k1 +1 1q
(3.25)
and for k2

q =q
k =k 1
k1 k =0
q k = q k1
1 1q
only for |q | < 1
(3.26)
back to rectangular pulse with (3.25) M (k ) RM (ej ) =

M 1 1 ej M j M2 sin( 2 ) = e 1 ej sin( ) 2
(3.27)
33
3.2
3.2.1
Z transforms
Denitions
v (k ), k Z, v C V (z ) =
k =
v (k ) z k = Spectrum
Z-Transform (3.28) inverse transformation V (z ), z C , V C v (k ) = Closed integration path in z-plane counter-clockwise. Again(see 3.1,3.2): (3.29)= spectral synthesis general complex exponentials z k , weighted by V (z ) = amplitude densities. (3.28)= spectral analysis i.e, determination of suitable weights. Remarks: (3.28-3.29) = generalization of (3.1-2) as z k generalizes ej k , regarding the unit z = ej in the z-plane from 3.28
+ 1 V 2j
1 2j
V (z ) z + k
dz z
(3.29)
, i.e. v (k ) as superposition of (now:) (z ) dz as complex z amplitudes
circle
V (z = e ) =
k =
v (k )ej k = F {v (k )}
3.1 dz d
and from (3.29) for a xed r , z = r ej v (k ) = 1 j 2

d rej , d
dz = jr ej d
V (z = ej ) ej k jej d
1 1 = j e 2
v (ej )ej k d

34
z = re j
z = 1 e j
Purpose of this generalization? Answer: Spectral representation of signals possible which are not F -Transformable (see condition 3.3)
3.2.2
Existence
v (k ) must be absolutely summable.
(see sec. 3.1.2): nite |v (z )|
| V (z )| = |
k = +
v (k )z k |
+ k =
|v (k )||z k |
=
k = +
|v (k )|r k |ej k |
1 ,k
| V (z )|
k =
| v (k )r k | M <
additional degree of freedom, choice of r = |z |
CHAPTER 3. TRANSFORM DOMAINS Still not every signal is allowed.

1
35
| V (z )|
k =
| v (k )r | +
k =0
| |
v (k ) | rk (3.30)
k =1
|v (k )r k | +
k =0
v (k ) | rk
There must exist values of r such that the summation in equation(3.30) is exactly summable, the values of r for which function is summable is called gence region of conver-
ROC. The region of convergence for the rst sum consists of all the points on a
circle of some radius r1 , where r1 < . For the second sum in the equation there must exist values of r large enough such that the product sequence v (k )/r k , 0 k < is absolutely summable. ROC of the second term consists of all points outside the circle of radius r > r2 . For general, two sided signals v (k ), convergence is needed for both k 0 and k 0. r 2 < | z | < r1 Consequences: r1 and r 2 < r1 r2 must exist. is necessary. (3.31)
Then: V (z ) exists according to (3.28) as the Z-transform of v (k ) in a ring of convergence i.e: for z-values in this ring acc. to (3.31) V (z ) corresponds to the sum (3.28) V (z ) may be evaluated at other points of the z-plane.
1111111111111 0000000000000 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111 0000000000000 1111111111111
Z (a)
(b)
ROC for (a) anticausal and (b) causal system
111111111111111 000000000000000 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 111111111111111 000000000000000 111111111111111 000000000000000
Z

111111111111111 000000000000000 Z 000000000000000 111111111111111 r 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 r 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111 000000000000000 111111111111111
1 2
36
ROC for a bidirectional signal
Example 1. Bi-directional sequence:
x(k ) =
k =0
ak u(k ) + bk u(k 1)
1
Solution: from (3.28) we have : X (z ) =

k =0
a z
k k
+
k =
b z
k k
=
k =0
(az ) +
l=1
1 k
(b1 z )l (3.32)
The rst term in the expression converges if |az 1 | < 1 term converges if |b1 z | < 1 or |z | < |b|
or |z | > |a|. The second
So in determining the region of convergence for X (z ) we have to consider two different cases. i. Case 1: In the case the two ROC above do not overlap, we can not nd values of z for which X (z ) converges. ii. Case 2: In this case there exists a ring in the z-plane, where both power series converge simultaneously.

11111111111111 00000000000000 00000000000000 11111111111111 |b| 0000000000000000 1111111111111111 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000000000 11111111111111 |a| 0000000000000000 1111111111111111 00000000 11111111 00000000000000 11111111111111 0000000000000000 1111111111111111 |b| 00000000 11111111 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000 11111111 |a| 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000 11111111 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000 11111111 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000 11111111 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000 11111111 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000 11111111 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000000000 11111111111111 0000000000000000 1111111111111111 This X (z) can exist 00000000000000 11111111111111 0000000000000000 1111111111111111 00000000000000 11111111111111 This X (z) can not exist (no overlap) Z
37
X (z ) =
1 1 1 1 az 1 bz 1 ba = a + b z abz 1 (3.33)
the ROC of X (z ) is |a| < |z | < |b|. 2. 1-sided sequence x(k ) = ak u(k ) X (z ) =
z
1 1 az 1
ROC:|Z | > |a|
the ROC in this case is would be one-sided.
3.2.3
Periodities
From (3.28), v (z ) is a-periodic in terms of general z-transform, however from z = rej .

+
V (z ) = V (re ) =
k =
v (k )r
k =+
=
k =
v (k )r k ej (+2)k
j 2 ej k e
1
= V (r e
j +2)
(3.34)
CHAPTER 3. TRANSFORM DOMAINS for any xed value of r . V (z ) is periodic with regard to the angle on a circle with radius r r = 1, unit circle!)
38
(which includes
3.2.4
Inverse Z-transform
Equation (3.29) denes the inversion of z -transform. v (k ) = 1 j 2 V (z )z k1 dz
The integral is a contour integral over a closed path C that encloses the origin and lies within region of convergence of V (z ). C can be taken as a circle in ROC of V (z ) in the z -plane. Possible methods of evaluating z -tranform are,
Direct evaluation of equation (3.29). Expansion into a series of terms, in variables z and z 1 . Partial-fraction expansion and table lookup.
Here only 3 is treated. A rational function for which the order of numerator polynomial M is greater then denominator N is called improper function i.e M > N . Any improper function can be made proper using long division. Partial-fraction expansion is particularly useful if V (z ) is a proper rational function and without loss of generality we consider that a0 = 1 V (z ) =
M k k =0 bk z N k k =0 ak z
b0 + b1 z 1 + b2 z 2 + + bM z M 1 + a1 z 1 + a2 z 2 + + bN z N
(3.35)
Since numerator and denominator are both polynomials in z , V (z ) can be expressed in factored form. The task is to perform partial-fraction expansion to express (3.35) as a
CHAPTER 3. TRANSFORM DOMAINS sum or product of simple fractions. V (z ) = (z z01 )(z z02 ) . . . (z z0M ) b0 N (z ) = z M +N D (z ) a0 (z z1 )(z z2 ) . . . (z zN )
39
(3.36)
This representation can be directly used to show V (z ) on a pole-zero plot. Pole zero plot and ROC information can completely describe the system features such as causality, stability, amplitude and phase characteristics. For examples (see exercises).
3.2.5
Characteristics rules,theorems
a. Linearity: as in the Fourier transforms see sec(3.1.4) b. Modulation: given: v (k ) V (z ) Generalized complex modulation
k v1 (k ) = v (k ) z0
then from (3.28) v1 (k ) V ( c. Shift: Given: v (k ) V (z ) Delayed / Advanced complex signal z ) z0 (3.37)
v1 (k ) = v (k k0 ) then from (3.28)
v1 (k ) V (z ) z k0 = factor ej k0 , i.e. Linear phase, if z = ej as in (3.9). a factor of z k0 describes a shift ( delay ) by k0 d. Convolution & Product: with denition of convolution in (3.15) we nd from (3.28) ( Shift operator
(3.38)
40
v1,2 (k ) Y (z )
V1,2 (z )
+ + n=
=
k =
v1 (n)v2 (k n) z k
v1 (k ) v2 (k ) = y (k ) Y (z ) = V1 (z ) V2 (z )=(3.17) for z = ej (3.39) ROC is atleast the intersection of V1 (z ) and V2 (z ) e. Dierence & Dierentiation: Given: v (k ) V (z ) using the shift theorem(3.38) Dierence: y (k ) = v (k ) v (k 1) Y (z ) = V (z )[1 z 1 ] which compounds to (3.9) for z = ej Dierentiation: z [ dV (z ) ] k v (k ) dz (3.41) (3.40)
One-sided z-Transforms
Primarily: Generally, two-sided denition(3.28) is applicable to one-sided signals ( with simpler convergence conditions ). Possible: Denition of a one-sided (right hand sided) transformation independently of signal values on the other side
Z R { v (k )} =
k =0
v (k )z k
(3.42)
with same inversion as in (3.29). Characteristics, rules and theorems remain, unchanged too - with simpler convergence consideration( circle rather then a ring). Additional
CHAPTER 3. TRANSFORM DOMAINS considerations with the shift theorem, in the case of shifts to the left. assumed : v (k ) 0 for k < 0 (Right handed sequence)
3.28
41
Z R { v (k )} = V R (z ) Z { v (k )} = V (z ) Shift: y (k ) = v (k + k 0 ) k 0 > 0
Y R (Z ) =
k =0
v (k + k 0 )z k =
v (k )z 0 + v (k0 + 1)z 1 + v (k0 + 2)z 2 + . . . 0 v (0)z k0 + v (1)z k0 1 + + v (k0 1)z k 0 1 k k
k=0
v(k )z
+k 0 k =0
v (k )z
VR (z )
k 0 1
k =0
v (k )z k 0 k
k 0 1
k0
V R (z ) z
k0 k =0
v (k )z k
3.3
3.3.1
Discrete Fourier transforms

Denitions
Denitions of discrete Fourier transforms

M 1
v (k ) VM () =
k =0
k v (k )WM
(3.43)
1 V M ( ) v (k ) = M where WM = ej 2/M , M is the no. of DFT points
M 1
k v (k )WM
(3.44)
=0
42
ej M
point DFT
3.3.2
Properties
a. Periodicity: If vM (k ) VM () then v (k + M ) = v (k ) k V ( + M ) = V ( )
This property is evident from (3.43) and (3.44). M point DFT of a nite length sequence v (k ) is equivalent to M point DFT of a periodic sequence vp (k ) with period M and, vp (k ) =
l=
v (k lM )
(3.45)
like DFS, DFT assumes the signal to innite and periodic, the coecients are calculated over one interval. Relation between DFS and DFT should be explored further.
CHAPTER 3. TRANSFORM DOMAINS b. Linearity: Linearity property holds for DFT: if v1 (k ) V1 () and v2 (k ) V2 () then v1 (k ) + v2 (k ) V1 () + V2 ()
43
(3.46)
If v1 (k ) has length M1 and v2 (k ) has length M2 , then maximum length of v (k ) is max[M1 , M2 ], which means shorter sequence must be appended with additional zeros ( Zero-padding ). c. Symmetry: Any given sequence can be decomposed into even and odd components, a sequence is said to be even: if v (k ) = v (k ) = v (M k ) odd: if v (k ) = v (k ) = v (M k ) If v (k ) C then v (k ) V (M ) and v (M k ) V ( ) A complex sequence v (k ) can be decomposed in to even and odd components veven (k ) = 1 [v (k ) + v (k )] 2 1 [v (k ) v (k )] vodd (k ) = 2 (3.47) (3.48)
For a real valued sequence v (M k ) = v (k ) = v (k ) |V (M )| = |V ()| and V (m ) = V ()
CHAPTER 3. TRANSFORM DOMAINS d. Shift & Modulation: if v (k ) V () then v (k l ) V ( )e and v (k )e e. Time reversal: if v (k ) V () then v (k ) = v (M k ) V (M )
M 1
j 2k0 M j 2l M
44
(3.49)
V ( 0 )
(3.50)
DF T {v (M k )} = change index from k to m = M k ,then =

m=0 M 1 k =0 M 1
v (M k )ej
2k M
v (m)e
2 (mM ) M
M 1
=
k =0
v (m)ej
2m M
V (M ) =
m=0
v (m)ej
2m(M ) M
(3.51)
3.3.3
Linear and circular convolution
linear convolution of two sequences v1 (k ) and v2 (k ),k Z yl (k ) = v1 (k ) v2 (k ) = v2 (k ) v1 (k )

+ +
=
=
v1 ()v2 (k ) =
=
v2 ()v1 (k )
(3.52)
circular convolution of two periodic sequences v1 (k ) and v2 (k ), k = {0, 1, . . . , M1,2 1} with the same period M1 = M2 = M and k0 Z yc (k ) = v1 (k )
k 0 +M 1
v2 (k ) = v2 (k ) v1 ()v2 (k ) =
v1 (k )
k 0 +M 1
=
=k 0
v2 ()v1 (k )
=k 0
(3.53)
45
DFT and circular convolution

Inverse transform of a nite-length sequence v (k ). k, = 0, . . . , M 1 v (k ) VM () v (k ) = v (k + M ) (3.54)
DFT of a nite-length sequence and its periodic extension is identical circular convolution property (k, = 0, . . . , M 1), for two nite length sequences v1 (k ) and v2 (k ).
y (k ) = v1 (k ) Proof:
v2 (k ) Y () = V1M () V2M ()
(3.55)
1 IDFT of y (k ) = M = Inserting DFT denition of V1M and V2M y (k ) = 1 M 1 M 1 M
M 1
k Y ()WM
=0 M 1 k V1M () V2M ()WM
=0
(3.56)
M 1 M 1 n v1 (n)WM M 1 M 1 l k WM v2 (l)WM
=0 M 1
n=0 M 1
l=0
v1 (n)
n=0 l=0
v2 (l)
=0
WM
(k nl)
(3.57)
Terms in bracket summation over the unit circle M 1 M f or l = k n + M, Z (k nl) WM = =0 0, otherwise Substituting (3.58) in (3.56) yields 1 y (k ) = M
M 1 +
(3.58)
v1 (n)
n=0 =
v2 (k n + M )
v2 ((k n))M (periodic extension)
M 1
=
n=0
v1 (n)v2 ((k n))M v2 (k )
(3.59) (3.60)
= v1 (k )
46
3.3.4
Frequency analysis of stationary signals
Spectral analysis of analog signal Antialiasing lowpass ltering and sampling with s m . For practical purposes (delay, complexity), limitation of the signal duration to the time interval T0 = LT (where L is the number of samples and T is sampling interval). Limitation of signal duration to T0 can be modeled as multiplication of input signal v (k ) with a rectangular window w (k ) 1 f or 0 k L 1 0, otherwise
v (k ) = v (k ) w (k ) with w (k ) =
(3.61)
Suppose that the input sequence just consists of a single sinusoid i.e. v (k ) = cos(0 k ), the Fourier transform is
V (ej ) = (0 ( 0 ) + 0 ( + 0 )) For window function w (k ) the Fourier transform can be obtained as

L1
(3.62)
W (e ) =
k =0
ej k =
L 1 sin( ) 1 ej L j L 2 2 = e ej sin( ) 2
(3.63)
We nally have
(ej ) = V
1 V (ej ) W (ej ) 2 1 = W (ej (0 ) ) + W (ej (+0 ) ) 2
(3.64)
47
14
L =25
12
10
Magnitude
0 Frequency
Windowed spectrum of V (ej ) is not localized to one frequency instead it is spread out to over whole frequency range. spectral leakage The rst zero-crossing of W (ej ) occurs at z = 2/L. The larger the number of sampling points L (thus also the width of rectangular smaller becomes z (and thus also the main lobe of spectrum). resolution leads to an increase of ). time resolution
window) the
Decreasing the
frequency
and vice versa ( duality of time and frequency
In practice DFT is used in order to obtain a sampled representation of spectrum according (ej ), = 0, . . . , M 1. to V Special case: M = L and 0 =
2 , M
= 0, . . . , M 1
Then the Fourier transform is exactly zero at the sampled frequencies except when = Example:M = 64 samples, k = 0, . . . , M 1 and rectangular window w (k ) . v0 (k ) = cos 5 2 2 k , v1 (k ) = cos 5 + k M M M (3.65)

v (k)=cos((5*2pi*/64+pi/64)k) 1
1
48
v(k)=cos(5*2pi*k/64)
0.5
0.5
0.5
0.5
20
40
60
20
k 40 DFT of 1 v (k)
60
DFT of v(k) 1 0.8 0.6 0.4 0.2 0 1 0.8 0.6 0.4 0.2 0
10
20
30
10
20
30
0 (ej ) = V0 (ej ) for v0 (k ) V
W (ej ) = 0 for = 2/M for = 5, since
0 is exactly an integer multiple of 2/M . Periodic repetition of v0 (k ) leads to a pure cosine sequence .
0 (ej ) = 0 for = 2/M , for v1 (k ) slight increase in 0 = 2/M for Z V periodic repetition is not a cosine sequence anymore.
49
Windowing with dierent window functions

Windowing not only distorts the reduces the spectral spectral estimate due to leakage eects, it also
resolution .
Consider a sequence of two frequency components v (k ) = cos(1 k ) + cos(2 k ) with the Fourier transform 1 W (ej (1 ) ) + W (ej (2 ) ) + W (ej (+1 ) ) + W (ej (+2 ) ) 2 where W (ej ) is the Fourier transform of the rectangular window from (3.63). 2/L < |1 2 | : Two maxima, main lobes for both window spectra W (ej (1 ) ) and W (ej (2 ) ) can be separated. 2/L = |1 2 | :Correct values of spectral samples , but main lobes can not be separated anymore. 2/L > |1 2 | :Main lobes of W (ej (1 ) ) and W (ej (2 ) ) overlap.
(3.66)
50
the ability to resolve dierent frequencies is limited by the main lobe width, which also depends on the length of the window impulse response L. Example: v (k ) = cos(0 k ) + cos(1 k ) + cos(2 k )
with 0 = 0.2, 1 = 0.22, 2 = 0.6 with window length (a)L = 25 (b)L = 50 and (c)L = 100. 0 and 1 can be separated only for window length of 100. Reduction of the sidelobe and reduced resolution compared the rectangular window can be clearly observed. Alternatives: dierent windowing functions Comparison of rectangular, hamming, hanning widows (L = 50)
51
Name of Window Rectangular Hamming Hanning Bartlett Blackman
Time Domain function w (k ), 0 k L 1 1

2k 0.54 0.46cos L 1 1 2 2k 1 cos L 1
1 2|k L | 2 L1
2k 4k 0.42 0.5cos L + 0.08cos L 1 1
Remark: Spectral analysis using DFT Sampling grid can be made arbitrarily ne by increasing the the length of windowed signal with zero padding (that is increasing M ). However, the spectral resolution is not increase (separation of two closely adjacent sine spectral lines). An increase in resolution can only be obtained by increasing the length of the input
CHAPTER 3. TRANSFORM DOMAINS signal to be analyzed (that is increasing L), which also results in a longer window.
52
CHAPTER 3. TRANSFORM DOMAINS frequency responses for dierent window lengths.
53
54
3.3.5
Ecient computation of DFT, FFT
Complexity of DFT calculation in (3.43) for v (k ) C, VM () C

M 1
V M ( ) =
k =0
k v (k )WM 1 complx multiplication
for : 0, . . . , M 1
M results
(3.67)
M complx mult., M cmplx add.
M2
overall complex multiplications and M 2
complex additions.
Remarks: 1 complex multiplication 4 real-valued mult.+2 real-valued additions .
1 complex addition is 2 real-valued additions. A closer evaluation reveals that there are less then M 2 operations: M values have to be added up (M 1) additions. Factors such as ej 0 , ej, ej 2 no real multiplications For = 0 no multiplications at all. Complexity of DFT becomes extremely large for large values of M (i.e. M=1024) ecient algorithms necessary for practical implementation. Fast algorithms for DFT, fast
k such as: Fourier transform (FFT) exploit symmetry and periodicity properties of WM
complex conjugate symmetry: WM
(M k ) (k +M )
= =
k WM
k = (WM ).
k Periodicity in and k : WM = WM
WM
(+M )k
3.3.6
Principle:
Decimation-in-time FFT algorithm
Decomposing the DFT computation into DFT computations of smaller size by means of decomposing M point input sequence of v (k ) into smaller sequences Decimation-in- time .
Perquisites:
CHAPTER 3. TRANSFORM DOMAINS M integer must be a power of 2 i.e.M = 2m , m = log2 (M ) N Radix-2
55
Decomposing M-point DFT into two M/2-point transforms DFT V ()(M is dropped for clarity) can be written as
M 1
V ( ) =
k =0
k v (k )WM , M/21
= 0, . . . , M 1
M/21
=
k =0
2k v (2k )WM
+
k =0
v (2k + 1)WM
(2k +1)
(3.68)
where in the last step v (k ) is separated into two M/2point sequences consisting of the even and odd numbered points in v (k ). since
2 = e2j 2/M = ej 2/(M/2) = WM/2 WM
we can write (3.68) as

M/21 M/21 k v (2k )WM/ 2 k =0
WM k =0
k v (2k + 1)WM/ 2
(3.69)
H ( ), = G() + WM
= 0, . . . , M 1
(3.70)
Each sum in (3.70) represents a M/2 point DFT over even and odd-numbered points of v (k ), respectively. G() and H () need to be computed over 0, . . . , M/2 1 points since both are periodic with period M/2. signal owgraph for M=8
56
v(0) v(2) v(4) v(6)
G(0)
0 WM
V (0) V (1)
1 WM
G(1)
M 2 point
DFT
G(2)
2 WM
V (2) V (3)
3 WM
G(3)
v(1) v(3) v(5) v(7)
H(0)
4 WM
V (4) V (5) V (6) V (7)
H(1)
M 2 point 5 WM
DFT
H(2)
6 WM
111 000
H(3)
7 WM
Complexity: 2 DFTs of length M/2 2 M 2 /4 = M 2 /2 operations + M operations for combining of both DFTs. M + M 2 /2 operations (less than M 2 for M > 2). Decomposition in 4 M/4-point DFTs G() and H () from (3.70) can be written as
M/41 M/41 k g (2k )WM/ 4 k =0
G ( ) =
WM/ 2 k =0
k g (2k + 1)WM/ 4
(3.71)
M/41
M/41 k h(2k )WM/ 4
H ( ) =
k =0
WM/ 2 k =0
k h(2k + 1)WM/ 4
(3.72)
where = 0, . . . , M/2 1.
57
v(0) v(4) v(2) v(6)
M 4 point
G(0)
DFT
W0 W1
M 2
G(1)
111 000 111 000
M 2
M 4 point
2 WM
2
G(2) G(3)
DFT
3 WM
2
The overall ow graph now looks like this
v(0) v(4) v(2) v(6)
M 4 point
V ( 0)
0 WM 2 WM 0 WM
DFT
V (1)
1 WM
111 000 111 000 111 000 111 000 111 000 111 000
M 4 point
4 WM 6 WM
V (2)
2 WM
DFT
V (3)
3 WM
v(1) v(7) v(3) v(5)
M 4 point
DFT
0 WM 2 WM
4 WM 5 WM 6 WM 7 WM
V (4) V (5) V (6) V (7)
M 4 point
4 WM 6 WM
DFT
Complexity: 4 DFTs of length M/4 M 2 /4 operations + 2 M/2 + M operations for reconstruction. M 2 /4 + 2M complex multiplications and additions. nal step: Decomposition of 2-point DFT
CHAPTER 3. TRANSFORM DOMAINS DFT of length 2:

0 2 (0) = v (0) + W2 v (1) = v (0) + v (1) V 1 2 (1) = v (0) + W2 v (1) = v (0) v (1) V
58
(3.73) (3.74)
Flow graph:
1 00 11 v (0) 0 00 1 11 1 0 000000000 111111111
v (1) 0 1
111111111 000000000 000000000 111111111 1 000000000 111111111 000000000 111111111 000000000 111111111 000000000 111111111 000000000 111111111 1 11 000000000 111111111 0 1 00 11 00 000000000 111111111 V2 (1) 0 1 00 11 00 11 1
1 0 V2 (0) 0 1
Insert this ow-graph from the last step in the above structure yields overall signal ow graph.
v(0)
0 WM 0 WM 2 WM 0 WM 4 WM 6 WM 0 W8
V (0) V (1)
1 W8
v(4) v(2) v(6)
4 WM
V (2)
2 W8
4 WM
V (3)
3 W8
v(1)
0 WM 0 WM 2 WM 4 WM 6 WM
4 WM 5 WM 6 WM 7 WM
V (4) V (5) V (6) V (7)
v(5) v(3)
4 WM
0 WM
v(7)
4 WM
In general, our decomposition requires m = log2 (M ) = ldM stages and for M
1 we have
M m = Mlog2 M
complex multiplications & additions instead of M 2 .
59
Examples M = 32 M 2 1000, M ld M 160 factor of 6 M = 1024 M 2 106 , M ld M 104 factor of 100 Buttery computation: Basic building block of above ow graph is called buttery 0, . . . , M/2 1
1 1 0 00 11 0 1 00 11 000000000 111111111 000000000 111111111 000000000 111111111 WM 000000000 111111111 000000000 111111111 000000000 111111111 000000000 111111111 000000000 111111111 1 000000000 111111111 0 1 0 1 000000000 111111111 0 1 0 1 M
(+ WM 2
11 00 00 11
Simplification
1 0 0 1
WM
1 1 0 00 11 0 1 00 11 000000000 111111111 000000000 111111111 1 000000000 111111111 000000000 111111111 000000000 111111111 000000000 111111111 000000000 111111111 1 000000000 111111111 000000000 111111111 0 1 0 1 000000000 111111111 0 1 0 1 1
the simplication is due to the fact that WM

M/2
= ej (2/M )M/2 = ej = 1 WM
+M/2 = WM WM M/2 = WM
Using these modications, the resulting ow graph for M = 8 is given as v (k ) values at the input of the decimation-time ow graph in permuted order.
Input data is stored in bit-reversed order.
60
Stage 1
Stage 2
Stage 3
v(0) v(4) v(2) v(6)

0 WM 0 WM
V (0) 1
0 WM
V (1) 1 1 W80 V (2) V (3)
2 WM
v(1) v(5) v(3) v(7)

0 WM 0 WM
1 1
0 WM
V (4) V (5) V (6) V (7)
W81 1 W82 1 1 W83 1 1
2 WM
Inverse FFT
according to (3.44) we have for the IDFT
1 v (k ) = M that is 1 v (k ) = M v (M k ) =
M 1
k VM ()WM
=0 M 1 k VM ()WM , =0
1 DF T {VM ()} M
(3.75)
with additional scaling and permutation the IDFT can be calculated with the same FFT algorithm as DFT.
61
Other alternatives
Decimation-in-frequency Radix-r and mixed Radix FFT Inplace computation
3.3.7
Transformation of real-valued sequences
v (k ) R FFT program/hardware vR (k ) + jvi (k ) C inecient due to performing arithmetic calculations with zero values.
=0
In the following we develop a method for ecient usage of a complex FFT for real-valued data.
FFT of two real-valued sequences

Given: v1 (k ), v2 (k ) R, k = 0, . . . , M 1 How can we obtain VM1 () v1 (k ), VM2 () v2 (k )? Dene: v (k ) = v1 (k ) + jv2 (k ) leading to the DFT (3.76)
VM () = DF T {v (k )} = VM1 () + jVM2 ()
C C
(3.77)
CHAPTER 3. TRANSFORM DOMAINS v (k ) + v (k ) 2 v (k ) v (k ) v2 (k ) = j2 v1 (k ) = Hence the DFT of these sequences is:
62
(3.78) (3.79)
1 [DF T {v (k )} + DF T {v (k )}] (3.80) 2 1 V 2 ( ) = [DF T {v (k )} DF T {v (k )}] (3.81) 2j V 1 ( ) = Separation of VM () into VM1 () and VM2 ()? v (k ) = vRe (k ) + vRo (k ) + j (vIe (k ) + vIo (k ))
v1 (k ) v2 (k )
(3.82)
DFT components v (k ) = vRe (k ) + vRo (k ) + jvIe (k ) + jvIo (k )
VM () = VMRe () + VMRo () + jVMIe () + jVMIo () Thus we have, V M 1 ( ) = where 1 j V M R ( ) + V M R (M ) + V M I ( ) V M I (M ) 2 2
(3.83)
(3.84)
VMRe () =
1 VMR () + VMR (M )) 2 1 (VMI () VMI (M ))) VMIo () = 2 (3.85)
likewise, we have for VM2 () the relation 1 j V M I ( ) + V M I (M ) + V M R ( ) V M R (M ) 2 2
V M 2 ( ) =
(3.86)
CHAPTER 3. TRANSFORM DOMAINS with 1 VMI () + VMI (M )) 2 1 (VMR () VMR (M ))) VMRo () = 2 VMIe () =
63
(3.87) Rearranging the results (3.84) and (3.86) VM 1 = VM 2 1 V M ( ) + V M ( M ) 2 j = V M ( ) V M ( M ) 2 (3.88) Due to Hermitian symmetry of real-valued sequences
(M ) VM(1,2) () = VM (1,2)
(3.89)
The values VM(1,2) () for { M + 1, . . . , M 1} can be obtained from those for 2 {0, . . . , M }such that only calculation of 2
M 2
+ 1 values in necessary.
Furthermore, DFT of 2M point real-valued sequences possible with a M point DFT.
Fast convolution of two real-valued sequences using DFT/FFT.
Chapter 4 Digital lters
4.1
Digital lter structures
Digital lter is a linear time-invariant (LTI) causal system with a rational transfer function. H (z ) =
N
bN i z i (4.1) zi aN i
i=0 N i=0
Where a0 = 1 without loss of generality. ai , bi : parameters of LTI system ( coecients of the digital lter ), N is the order of the lter. 4.1 in product notation.
N
H (z ) =
=0 b0 iN i=0
(z z0i ) (4.2)
(z zi )
where the z0i are the zeros, and zi are the poles of the transfer function. In a lter poles are responsible for stability. Equation 4.1 can also be expressed as a dierence equation
N N
y (k ) =
i=0
bi v (k i)
i=1
ai y (k i)
(4.3)
64
CHAPTER 4. DIGITAL FILTERS
65
with v (k ) denoting the input signal and y(k) denoting the resulting signal after ltering. Generally 4.3 describes the innite impulse response ( IIR ) lter. and recursively from y (k
y (k ) is calculated from v (k ), v (k 1), . . . , v (k N ) 1) , y (k 2) , . . . , y (k N ).
The calculation of y (k ) requires memory elements in order to store
v (k 1) , . . . , v (k
N ) and y (k 1) , . . . , y (k N ) we have a dynamical system. bi 0 for i = 0. H (z ) = b0 z N

N
= zi
b0 z N
N i=1
(4.4)
aN i
i=0
(z zi )
this lter has no zeros All-pole or auto-regressive (AR)-lter. The transfer function is purely recursive.
N
y (k ) = b0 v (k )
i=1
ai y (k i)
(4.5)
ai 0 for all i = 0, a0 = 1(causal lter required): Dierence equation is purely
non-recursive.
N
y (k ) =
i=0
bi v (k i)
(4.6)
Non-recursive lter. Transfer function: 1 H (z ) = N z Remarks: Poles zi = 0, i = 1, , N , but not relevant for stability all-zero lter According to (4.6), y (k ) obtained by weighted average of the last N + 1 input values Moving average (MA) lter, as opposite to AR lter from above.
N N
bN i z =
i=0 i=0
bi z i
(4.7)
From (4.7) it can be seen that the impulse response has a nite length Finite impulse response (FIR) lter of length L=N +1 and order N .
66
4.1.1

Structures of FIR lters
Dierence equation given by (4.6). Transfer function given by (4.7). Unit impulse response is equal to the coecients bi . bk 0 0k L1 otherwise,
h(k ) =
Direct form Structure

The direct form structure follows immediately from non recursive dierence equation given in (4.6), which is equivalent to the linear convolution sum
L1
y (k ) =
n=0
h(n)v (k n)
(4.8)
v(k)
z1
h(0)
z1
h(1)
z1
z1
h(L 3) h(L 2) h(L 1) y(k)
tapped-delay-line or transversal lter in rst-direct form. If the unit impulse v (k ) = 0 (k ) is chosen as input signal, all the samples of the impulse response h(k ) appears successively at the output of the structure. Second direct-form can be obtained by transposing the ow graphs. i.e. Reversing the direction of all branches. Exchanging the input and output of the ow-graph. Exchanging the summation points with the branching points and vice versa.
67
v(k) h(L 1) h(L 2) h(2) h(1) h(0) y(k)
z1
z1
z1
Linear-phase lters
If the impulse response of a causal FIR lter satises either the symmetry or asymmetry condition. h(k ) = h(L 1 k ) Linear-Phase lter. Therefore in cases of symmetry/asymmetry in impulse response the number of multiplications required reduce from L to L/2 if L is even and (L + 1)/2 for L is odd. (4.9)
v(k)
z 1
z 1
z 1
z 1
z 1
z 1
h(0) h(1)
z 1
z 1
z 1
z 1
h( L 2 2) h( L 2 1)
h( L 2 3)
y(k)
Exercise: Construct the diagram for odd L. we dene s = (L 1)/2.

L1
68
H (z ) =
k =0 s 1
h(k )z k h(k )z k
s 1 s 1
=
k =0
h(k )z k2s
s 1
k =0
= z
h(k )z
k =0 s 1
s k
k =0
h(k )z ks (4.10)
H (z ) = z s The frequency response
h(k ) z (sk) z (sk)
k =0
H (ej ) = ejs 2
L/21
h(k )cos ((s k ))
For even L, even symmetry (4.11) For even L, odd symmetry (4.12)
k =0
H (e ) = j e
j js
L/21
h(k )sin ((s k ))
k =0
The responses for odd length sequences can be calculated similarly. The Z-transform of equation (4.9) is z (L1) H (z 1 ) = H (z ) (4.13)
1 Equation (4.13) implies that if zo is root of the polynomial H (z ) then z0 , z0 and 1/z0
are also roots of the polynomial. The roots of linear phase lter occur in quadripples.
1 z 1 1 z 3
z3
z1
z2 z 3
1 z3
1 z2
z 1
1 z1
69
Cascade form Structure

By factorizing the transfer function
C
H (z ) = H 0
c=1
H c (z )
(4.14)
we obtain a cascade realization. The Hc (z ) are normally second-order, since in order to

obtain real coecients, conjugate complex zeros z0i and z0 have to be grouped. i 1 z ) Hc (z ) = (1 z0i z 1 )(1 z0 i
= 1 + b1 z 1 + b2 z 2
(4.15)
for linear phase lters due to special symmetry (4.9) the zeros appear in quadripples.
mg
4.1.2
Structures of IIR lter
Direct form structure

Rational system function (4.1) can be viewed as two systems in cascade H (z ) = N (z )/D(z ) = H1 (z ) H2 (z ) with
N
H 1 (z ) =
i=0
bi z i ,
H 2 (z ) = 1+
1
N i=1
ai z i
70
The all-zero H1 (z ) can be realized with the direct form structure (4.1.1) by attaching the all-pole system H2 (z ) in cascade, we obtain direct form I realization.
v(k)
z1 z1
b0 b1 b2 bM1 a1 a2 aN 1 aN
z1 z1
y(k)
z1
z1
bM
This structure requires M + N + 1 multiplications, M + N additions and M + N + 1 storage locations. Where M and N represent the order of numerator and denominator polynomials. Another realization can be obtained by exchanging the order of the all-pole and all-zero lter. Then, the dierence equation for the all-pole section is
N
w (k ) =
i=1
ai w (k i) + v (k ) b0
where w (k ) is an intermediate result and represents the input to the all-zero section:
N
y (k ) =
i=0
bk w (k i)
The resulting structure is given as follows
71
v(k) a1 a2 aN 1 aN
z1 z1 z1
b0 b1 b2 bN 1 bN
y(k)
only a single delay line is required for storing the delayed versions of the sequence w (k ). The resulting structure is called direct form II realization. Furthermore, it is said to be canonic, since it minimizes the number of memory locations(among other structures). Transposing the direct form II realization leads to the following structure, which requires the same number of multiplications, additions and memory locations.
v(k)
b0 b1 b2 bN 1 bN
z1 z1 z1
y(k) a1 a2 aN 1 aN
72
Cascade form structure

Like section(4.1.1) we can also factor an IIR lter system H (z ) into rst and second order subsystems Hp (z ) according to
N
H (z ) =
p=1
H p (z )
(4.16)
Degree of freedom in grouping poles and zeros. First order system canonical direct form for N = 1
v(k) a1
z 1
b0 b1
y(k)
H (z ) =
Y (z ) b0 + b1 z 1 = V (z ) 1 + a1 z 1
(4.17)
every rst order transfer function can be realized with the above ow-graph. As explained before a0 can be set to 1 without loss of generality. Second order system canonical direct form for N = 2
v(k) a1 a2
z 1
b0 b1 b2
y(k)
z 1
H (z ) =
b0 + b1 z 1 + b2 z 2 Y (z ) = V (z ) 1 + a1 z 1 + a2 z 2
(4.18)
73
Parallel form structure

an alternative to the factorization of a general transfer function is to use a partial fraction expansion, which leads to a parallel form structure. In the following we assume that we have only distinct poles. The partial fraction
expansion of a transfer function H (z ) with the numerator and denominator degree N is given as:
N
H (z ) = A0 +
i=1
Ai 1 zi z 1
(4.19)
A0
H1 (z) V (z) H2(z)
H p(z)
Y (z)
The Ai , i 1, . . . , N are the co-ecients(residues) in the partial fraction expansion, A0 = bn /an . Further more we assume that we have only real-valued coecients, such that we can
combine pairs of complex conjugate poles to form a second order subsystem i {1, .., N }. = A Ai i + 1 1 zi z 1 1 z iz
1 2 {Ai } 2 {Ai z b0 + b1 z 1 i }z = 1 {zi }z 1 + |zi|2 z 2 1 + a1 z 1 + a2 z 2
(4.20)
74
Two real-valued poles can also be combined to a second order transfer function
i, j {1, . . . , N } = = Aj Ai + 1 1 1 zi z 1 z jz (Ai + Aj ) (Ai zj + Aj zi )z 1 b0 + b1 z 1 = 1 (zj + zi )z 1 + (zi zj )z 2 1 + a1 z 1 + a2 z 2 (4.21)
If N is odd, there is real-valued pole left, which leads to one rst order partial fraction Example: Determine the cascade and parallel realization for the system described by the system function.
H (z ) =
z 1 )(1 2 z 1 )(1 + 2z 1 ) 10(1 1 2 3 (1 3 z 1 )(1 1 z 1 )[1 ( 1 + j1 )z 1 ][1 ( 1 j1 )z 1 ] 4 8 2 2 2 2
Solution in cascade form can be easily obtained by pairing the poles and zeros together, one possible solution is: z 1 1 2 3 H 1 (z ) = 3 2 1 7 z 1 + 32 z 8 3 1 2 1 + 2z z H 2 (z ) = 1 z 1 + 1 z 2 2 the overall lter is H (z ) = 10H1(z )H2 (z ) To obtain the parallel form representation H (z ) must be expanded in partial fraction. We have H (z ) = A2 A3 A A1 3 + + + 1 1 1 1 1 1 1 1 3 z 1 z 1 ( + j ) z 1 ( j1 )z 1 4 8 2 2 2 2
Where Ai are partial fraction coecients to be calculated. A1 = 2.93, A2 = 17.68, A3 = 12.25 j 14.57, A 3 = 12.25 + j 14.57. Combining the pairs of poles we have H (z ) = 14.75 + 12.90z 1 24.5 + 26.82z 1 3 2 + 1 7 z 1 + 32 z 1 z 1 + 1 z 2 8 2 (4.22)

10
75
v(k) z1
7 8
v(k)
z1
2 3 1
3 2
z1
3 32
z1
2 3
24.5 10
z1
1 26.82
z1 v(k)
2 3
y(k)
14.75
z1
7 8
12.9
z1
3 32
if H (ej ) exists it means the lter is stable and frequency response can be expressed as, H (ej ) = HR (ej ) + HI (ej ) (4.23)
If h(k ) is a real-valued and causal lter then from the symmetry properties of the Fourier transform in (3.12) imply that he (k ) HR (ej ) (4.24)
ho (k ) HI (ej )
(4.25)
So if h(k ) is completely specied by he (k ) then it follows that H (ej ) is completely specied by HR (ej ) and if h(k ) can be completely expressed in terms of ho (k ) then
CHAPTER 4. DIGITAL FILTERS H (ej ) can be completely found in terms of HI (ej ).
76
The magnitude and phase response of a causal lter are interdependent on each other and hence can not be specied independently.
4.2
Coecient quantization and roundo eects
The accuracy of digital systems is limited by their resolution. In this section we discuss the eects of xed-point digital lter implementation on the system performance.
Numerical representation in xed format:
A real number can be represented as:

B
= [A , . . . , 1 , 0 , . . . , B ] =
l =A
l r l
(4.26)
Where l is the digit r is the radix (base), A the number of integer digits,B the number of fractional digits. Example[101.01]2 = 1 22 + 0 21 + 1 20 + 0 21 + 1 22 Most important in digital signal processing: Binary representation with r = 2 and l {0, 1}, A MSB, B LSB b bit fraction format A = 0, B = b 1, binary point between 0 and l number
between 0 and 2 2b+1 are possible. positive numbers are represented as

b1
v = [0.1 2 . . . b 1] =
l=1
2 l
(4.27)
Negative fraction: v = [0.1 2 . . . b 1] =
b1
2 l
(4.28)
l=1
CHAPTER 4. DIGITAL FILTERS can be represented with one of the following three formats Sign-magnitude format:
77
vSM = [1.1 2 . . . b1 ] f or < 0 Ones Complement format:
(4.29)
v1C = [1. 1 2 . . . b1 ] f or Alternative denition:

b1
v<0
(4.30)
v1C = 1 2 +
0 l=1
(1 l ) 2l = 2 2b+1 |v |
(4.31)
Twos Complement format:
v2C = [1. 1 2 . . . b1 ] [00 . . . 01] f or
v<0
(4.32)
where denotes a modulo-2 addition (XOR). We thus have using 4.31 have v2C = V1C + 2b+1 = 2 |v | (4.33)
Example: Express fractions 7/8 and 7/8 in sign-magnitude, twos complement and ones complement format: v = 7/8 can be expressed as 21 + 22 + 23 , such that v = [0.111]. In sign-magnitude format, v = 7/8 is represented as vSM = [1.111], in ones complement code we have v1C = [0.111] and in twos complement the result v2C = [1.000] [0.001] = [1.001]. Remarks: Most DSPs use twos complement arithmetic. Thus any bbits can be viewed modulo
2b (Example for b = 3)
78
0 1
111 000 001 010 011 100
0 1 2
0.25 111 0.5 110 101 0.75 000 001 010 011 100 1.0 0.75 0.5 0.25
110 101
If the sum is within the range it will be computed correctly, even the sum of individual
partial sums result in overow.
Truncation and rounding Problem: Multiplication of the two bbit numbers yield a result of length 2b 1 truncation/rounding is necessary can be again regarded as quantization of the lter coecients v . Suppose that we have a xed-point realization in which a number v is quantized from bu to b bits. We rst consider the truncation case. Let the truncation error be dened as Et = Qt [v ]v . for positive numbers the error is (2b+1 2bu +1 ) Et 0 truncation leads to a number smaller then the unquantized number. for negative numbers and the sign-magnitude representation the error is 0 Et (2b+1 2bu +1 ) (truncation reduces the magnitude of the number) For negative numbers in the twos complement case the error is (2b+1 2bu +1 ) Et 0 (4.36) (4.35) (4.34)
79
(negative of a number is obtained by subtracting the corresponding positive number from 2 as in (4.33)). Quantization characteristic function for a continuous input signal.
Sign magnitude
Twos complement
Qt [v]
Qt [v]
2b+1 2b+1
2b+1
2b+1
Et = Qt (v) v
Et = Qt (v) v
Rounding case, rounding error is dened Er = Qr [v ] v : Rounding aects only the magnitude of the number and is thus independent from the
type of xed-point realization. Rounding error is symmetric around zero and falls in the range. 1 1 (2b+1 2bu +1 ) Er (2b+1 2bu +1 ) 2 2 Quantization characteristic function bu (4.37)
80
Q r [ v]
2b+1
2b+1 2
E r = Q r ( v) v
Numerical Overow:
If a number is larger/smaller then the maximal/minimal possible number representation (1 2b+1 ) for sign magnitude and ones complement arithmetic. 1 and (1 2b+1 ), respectively for twos complement arithmetic, we speak of an overow/underow condition. Overow example in twos complement arithmetic range{8, . . . , 7} [0.111] [0.001] = [1.000]
7 1 8
resulting error can be very large when an overow/underow occurs Twos complement quantizer for b = 3, = 2b
81
011 010 001
v = Q[v]
010 001 000
011
9 2
72
52
111
3 2
5 2
7 2
110 101 100 100 101
Alternative: Saturation or clipping, error does not increase abruptly in magnitude when overow/underow occurs: Disadvantage:Summation property of the twos complement representation is violated.
Q[v]
010 001 000
92 7 2 5 2
011
111
3 2
5 2
7 2
110 101 100
4.2.1
Coecient quantization
In a DSP/hardware realization of an FIR/IIR lter the accuracy is limited by the
word length of the computer Coecients have to be quantized. Word-length reduction of the coecients lead to dierent poles and zeros compared
CHAPTER 4. DIGITAL FILTERS to desired ones. This may lead to modied frequency response with decreased selectivity stability problems
82
sensitivity of quantization of lter coecients
Direct-form realization, quantized coecients ai = ai + ai bi = bi + bi where ai and bi represent the quantization errors. Example Deviation of Filter characteristics zi = zi z i when coecients ai are quantized (z i denotes the resulting pole after quantization). It can be shown that this deviation can be expressed as,
N
zi =
n=1
N n z i N
an
i = 1, . . . , N
(4.38)
l=1,l=i
(zi zl )
from (4.38) we can observe the following By using the direct form, each single pole deviation zi depends on all quantized
denominator coecients ai . The error zi can be minimized by maximizing the distance between |zi zl |
between the poles zi and zl . Splitting the lter into single or double pole sections (rst or second order transfer functions).
into a second order section leads to a small perturCombining the poles zi and z i
bation error zi since complex conjugate poles are normally suciently far apart. Realization in cascade or parallel form. The error of a particular pole pair zi and
is independent of its distance from the other poles of the transfer function z i
CHAPTER 4. DIGITAL FILTERS Example: Eects of Filter coecient quantization Consider the eects of coecient quantization for a bandpass IIR elliptic lter.
83
a. Magnitude frequency response 20log10 |H (ej )|(direct-form realization Unquantized) b. Passband details (direct-form realization Unquantized) c. Passband details for cascade structure.(16 bit quantization) d. Passband details for parallel structure.(16 bit quantization) e. Magnitude frequency response (log) for direct structure(16 bit quantization)
CHAPTER 4. DIGITAL FILTERS Cascade or parallel form?
84
Cascade form
H (z ) =
bp0 + bp1 z 1 + bp0 z 2 a + ap 1z 1 + ap0 z 2 p=1 p0

P
Parallel form
H (z ) = A0 +
p=1
cp 0 + cp 1 z 1 ap0 + ap1 z 1 + ap0 z 2
Cascade form: Only the numerator coecients bp i of an individual section determine
the perturbation of the corresponding zero locations(an equation similar to (4.38)) can be derived direct control over poles and zeros. Parallel form: A particular zero is aected by the quantization errors in the numer-
ator and the denominator coecients of all individual sections coecients cp0 and cp1 do not specify the position of a zero directly, direct control over poles only. Cascade structures is more robust against coecient quantization and should be used in most cases. Example: Elliptic lter of order 7
85
CHAPTER 4. DIGITAL FILTERS Coecient quantization in FIR systems
86
In FIR lters we only have to deal with locations of the zeros, since for causal lters all poles are at z = 0. Remarks: For FIR lters an expression analogous to (4.38) can be derived for the zeros FIR
lters also should be in cascade according to

P
H (z ) = H 0
p=1
(1 + bp1 z 1 + bp2 z 2 )
with second order subsections, in order to limit the eects of coecient quantization to the zeros of the actual subsection only. However,since the zeros are more or less uniformly spread in the z-plane, in many
cases the direct form is also used with quantized coecients. For a linear-phase lter with symmetry in the impulse response, the quantization does
not aect the phase characteristics, but only magnitude.
87
4.2.2
Round-o eects
Direct form-I structure
Direct form-I structure dierence equation

N N
y (k ) =
i=0
bi v (k i)
i=1
ai y (k i)
All signal values and coecients are represented by bbit binary xed-point numbers (for example twos complement representation) truncation or rounding of (2b 1)bit products to bbit necessary. modeling as constant real valued multiplication followed by a quantizer

b0
88
v(k) z 1
z 1
b1 a1
y(k)
z 1
b2 a2
z 1
v(k) z 1
b0
Q Q Q
Q
z 1
y(k)
a1
b1
Q Q
z 1
b2 a2
z 1
v(k) z 1
b0
y(k)
e0 (k)
z 1
a1
b1
z 1
b2
e1 (k)
e3 (k)
z 1
a2
e2 (k)
e4 (k)
This can be described as dierence equation

N N
y (k ) =
i=0
Q[bi v (k i)]
i=1
Q[ai y (k i)]
As we know from chapter 2 that the result of each single quantization stage can be modeled by adding a noise source ei (k ) with the following properties: Each ei (k ) corresponds to a wide-sense stationary white-noise process. Each ei (k ) has an uniform distribution of amplitudes over one quantization interval
i.e.(uniform pdf) Each ei (k ) is uncorrelated with the quantizer input, all other quantization noise
CHAPTER 4. DIGITAL FILTERS sources and the input signal of the lter. For bbit quantization the rounding error falls in the range 1 1 (2b+1 ) ei (k ) (2b+1 ) 2 2 and for twos complement truncation we have 2b+1 ei (k ) 0 mean and variance for rounding, = 0,
2 e =
89
22b+2 12
(4.39)
mean and variance for truncation, = 22b+1 , 2

2 e =
22b+2 12
(4.40)
For a second order lter the following structure can be obtained by summing
up all the noise sources:

4
e(k ) =
i=0
ei (k )
e(k) v(k) z 1
b1 a1 b0
z 1 z 1
b2 a2
y (k) = y(k) + f (k)
z 1
Overall noise variance in the general case: 22b+2 12
2 e = (2N + 1)
(4.41)
90
Due to linearity the output of the whole lter y (k ) = y (k )+ f (k ). Thus, the dierence
equation for the quantization noise e(k ) is given as

N
f (k ) =
i=0
ai f (k i) + e(k )
since e(k ) can be regarded as the input to all-pole system with the output f (k ). If the transfer function of the all-pole lter is dened as Hef (z ) = mean of f (k ):f = e Hef (ej ) = 0 for rounding.
2 |H (ej )|2 power spectral density f f (ej ) = e 2 1 variance f = F {f f (ej )}|k=0 + 1 D (z )
then
2 f
2 = e 2
|Hef (e )| d =
j 2
2 e k =
|hef (k )|2
(4.42)
(Parsevals theorem) Combining (4.41) and (4.42) the total noise variance due to internal round-o is
2 = (2N + 1) f
22b+2 2 12 2
2b+2
d |D (ej )| |hef (k )|2 (4.43)
= (2N + 1)
12
k =
Example Round-o Noise in rst-order system Given transfer function H (z ) = impulse response h(k ) = b0 ak 1 1 (k ) b0 , |a0 | < 1 1 a0 z 1
e(k) = eb0 (k)+ ea1 (k)
v(k) b0 a1 z 1
y(k)
CHAPTER 4. DIGITAL FILTERS Considering (4.43) and the two error sources ea1 (k ) and eb0 (k ), we have
2 f
91
22b+2 =2 12
|a1 |2k =
k =0
22b+2 1 ) ( 6 1 |a1 |2
(4.44)
The output of noise variance increases when the poles z = a1 approaches the unit circle In order to keep the noise variance below certain level wordlength b must be increased.
4.3
Computation of frequency response
In evaluating the magnitude response and the phase response as a function of frequency. Its convenient to express H (ej ) in terms of its poles and zeros.
M
H (ej ) = equivalently
=0 b0 iN i=1
(1 z0i ej )
(1 zi ej )
M
H (ej ) = b0 ej (N M )
i=0 N i=1
(ej z0i ) (4.45) zi ) (4.46)
(ej
expressing the complex-valued factors in (4.45) ej z0i = Vi (ej )eji () and ej zi = Ui (ej )eji () here Vi (ej ) = |ej z0i | Ui (ej ) = |ej zi | and and i (ej ) = (ej z0i ) i (ej ) = (ej zi )
The magnitude response of H (ej ) is equal to the product of magnitudes of all terms in (4.45) |H (ej )| = |b0 | V1 (ej ) . . . VN (ej ) U1 (ej ) . . . UM (ej ) (4.47)
CHAPTER 4. DIGITAL FILTERS the phase response
92
H (ej ) = b0 + (N M ) + 1 () + 2 () + + N () 1 () + 2 () + + 1 () (4.48)
L
pk
A x
e j
zk
L
pk
Vk
Uk
x
k ()
k ()
zk
93
4.4
Characteristics of practical frequency selective lters
As it is well-known that ideal lters are non-causal and hence practically unrealizable for real-time signal processing. Causality implies that the lter response characteristic H (ej ) can not be zero except for a nite set of points in the frequency range. In addition H (ej ) can not have innitely sharp cuto from pass-band to stop-band. Although ideal lter responses are desirable but they are not really required in practice. In particular its not necessary to insist on constant |H (ej )| in the pass-band and similarly it is not necessary to have H (ej ) as exactly zero in the stop-band. Small nonzero ripples in stop and pass band are tolerable.
Passband ripples
1 + 1 1 1
Transition Band
Stopband
The transition from passband to stopband denes the 1 passband ripple. 2 stopband ripple. p Passband edge frequency. s Stopband edge frequency. Filter design problem:
transition band
Specify 1 ,2 ,p and s corresponding to the desired application.
94
Select coecients ai and bi as free parameters such that the resulting frequency
response H (ej ) best satises the requirement of 1 ,2 ,p and s . The degree to which H (ej ) approximates the specication depends on the criterion
used for selecting ai and bi and also the order of polynomials, i.e the number of coecients. The impulse response of FIR lters is dened in (4.6) and (4.7). The specication of a lter can be dened in two ways absolute specication which provides a set of requirements on the magnitude response function |H (ej )|. The second approach is called relative specications, they provide the requirements in decibel[dB] H (ej )|max 0 dB scale = 20log10 H (ej ) The typical specications for lowpass lter are shown in gure. The band of frequencies that is allowed to pass through the lter is called the passband and is given by 0 || p . The band of frequencies that is suppressed by the lter is called stopband and is given by s . The band p s is called transition band. In absolute specications, we require that the passband and the stopband ripples 1 and 2 respectively must satisfy Passband: 1 1 |H (ej )| 1 + 1 , Stopband: |H (ej )| 2 for || p
for s ||
In the relative specications, we require that the passband ripple Rp and the stopband attenuation As must satisfy Passband: 0 dB scale Rp , Stopband: dB scale > As for || p
for s ||
The relationship between the two sets of parameters is given by Rp = 20log10 1 1 1 + 1 2 As = 20log10 1 + 1 (4.49) (4.50)
95
4.4.1
Design of FIR lters
Windowing functions
The natural approach to designing FIR lters via the window design is to choose a proper ideal (or desired) frequency selective lter (which is always noncausal,innite duration impulse response hd (k )) and truncate it to obtain a linear-phase and causal FIR lter. The impulse response of designed lter is given by: h(k ) = hd (k ).w (k ) (4.51)
where w (k ) is some symmetrical function over 0 k L 1 The approach of this method is to select suitable frequency response and windowing function. The frequency response of designed lter is H (ej ) = Hd (ej ) W (ej ) = 1 2

Hd (ej )W (ej () )d
(4.52)
(??) produces a smeared version of ideal lter. Example: An ideal lowpass lter with cuto frequency 0 < c < is dened as 1 ej (L1)/2 || c j HLP (e ) = 0 , c The impulse response of this lter is noncausal and of innite duration . Which can be truncated and shifted to yield a realizable lter. hLP (k ) =
1 )) sin(c (k L 2 1 (k L ) 2
(4.53)
96
Kaiser has formula to estimate the order of FIR lter N= 20log1 + 0.22 (4.54)
where = s p in digital specication.
4.4.2
Design of IIR lters
To design analog lters the magnitude-squared response should be provided. Three designs all widely used for prototyping lters, namely Butterworth, Chebyshev and Elliptic lters. The lter design requirements are dened as follows 1 1+ |H (j )|2 1, 1 , A2 | | p s | | (4.55)
0 |H (j )|2 where
is the passband ripple parameter,p is the passband cuto frequency, A is the
stopband attenuation parameter and s is stopband cuto frequency.The parameters are associated with relative parameters Rp and A by Rp = 10log10 1 1+ = 10Rp /10 1 (4.56) (4.57)
As = 10log10
1 A = 10As /20 A2
CHAPTER 4. DIGITAL FILTERS The squared magnitude response |H (j )|2 can be rewritten as |H (j )|2|s=j = H (s)H (s)
97
(4.58)
From here we can obtain the system function Ha (s) of analog lter. To obtain a stable and causal lter the left half poles and zeros of H (s)H (s) are assigned to Ha (s). Butterworth lters have monotone behavior both passband and stopband. The magnitudesquared response of an N-th order Butterworth lter is |H (j )|2 = 1 2N 1 + ( ) c (4.59)
Where N is the order of the lter and c is the 3dB frequency(cuto frequency). Using (4.58) in (4.59) H (s) H (s) = Poles of H (s) H (s): s2 = (1)1/N = ej (2n+1)/N 2 c s = c ej/2 ej (2n+1)/2N , n = 0, . . . , 2N 1 (4.60) 1 s2 N 1 + ( ) 2
c
Butterworth lter with N = 4
Poles of H ( s)
+ 8
28
Poles of H (s)
The 2N poles of H (s)H (s) occur on a circle of radius c at equally spaced points
in the S-plane.
98
N poles for n = 0 . . . , N 1 in (4.60) are located in the left half of the s-plane and
belong to H (s). The remaining poles lie in the right half of the s-plane and belong to H (s). Butterworth lter has N zeros at .
Estimation of required lter order N . At the stopband edge frequency (4.59) can be written as 1 2N 1 + ( ) c log ( 12 1)
2 s 2log ( ) c
2 = 2
which can be resolved to N=
(4.61)
Chebyshev Filter
Two type of Chebyshev lters, Type 1 lters are all pole lters with equi-ripple behavior in passband and monotonic
characteristic (similar to Butterworth lter) in the stopband. |H (j )|2 = 1 1+

2 T 2 (/ ) p N
(4.62)
CHAPTER 4. DIGITAL FILTERS Where TN (x) is the Chebyshev polynomial of N th order.
99
Type 2 Filters have poles and zeros, equi-ripple behavior in the stopband and mono-
tonic characteristics in passband. |H (j )|2 =

|H ( j )|2
1 1+2
1 1+
2 T 2 ( / )/T 2 ( / ) s p s N N
(4.63)
|H ( j )|2
Type1 (N odd)
Type2 (N odd)
ps
Estimation of lter order: Chebyshev lter only depends on the parameters N , ,2 and the ratio s /p . It can be shown that the required order can be estimated as N= log
2 1 2 + 2 1 2 (1 2 )/ (
2 )
log s /p +
(s /p )2 1
(4.64)
Elliptic Filter
Squared magnitude response: |H (j )|2 = 1 2 1 + 2 UN (/c) (4.65)
Where N is the order of Jacobian polynomial UN () and is the passband ripple. Elliptic lters provide very sharp magnitude response.
Butterwoth lters exhibit fairly linear phase response. Elliptic lters have highly nonlinear phase response and the phase characteristics of chebyshev lters lie in between.
100
4.5
Analog to digital transformation methods
Impulse invariance methods

The goal is to design a IIR lter with an impulse response h(k ) being a sampled version of the impulse ha (t) of the analog lter: h(k ) = ha (kT ), k = 0, 1, 2, . . .
The frequency response of the ideally sampled signal from (2.16) 1 H (j ) = T
Ha (j j
k =
2k ) T
(4.66)
T Should be chosen suciently small to avoid aliasing. Method is not suitable for design of highpass lters due to large amount of possible
aliasing.
Ha ( j)
H (e j )
Suppose that the poles of the analog lter are distinct. Then the partial fraction expansion of Ha (s) writes
N
H a (s ) =
i=1
Ai s s i
(4.67)
101
Where Ai are the coecient of partial fraction expansion, and si denote the poles of the analog lter. Inverse Laplace transform of (4.67) yields
N
ha (t) =
i=1
Ai esi t ,
t0
periodic sampling of ha (t)

N
h(k ) = ha (kT ) =
i=1
Ai esi kT
taking z transform of h(k )
H (z ) =
k =0
h(k )z
=
k =0 i=1
Ai esi kT z k
(4.68)
Then we have
N
H (z ) =
i=1
Ai
k =0
(e
s i T 1 k
z ) =
i=1
Ai s e i T z 1
Thus given an analog lter Ha (s) with poles si, the transfer function of the corresponding digital lter using impulse invariance transform is
N
H (z ) =
i=1
Ai s e i T z 1
(4.69)
Example: Convert the analog bandpass lter with system function. H a (s ) = into digital IIR lter. We note that the analog lter has zero at s = 0.1 and a pair of complex conjugate poles at s = 0.1 j 3. We do not have to nd the impulse response of ha (t) to sample it, instead we can directly determine H (z ) using (4.69) H (s ) = H (s ) = s + 0.1 j 3 1
1 2
s + 0.1 (s + 0.1)2 + 9
1 2
s + 0.1 + j 3 + 1
1 2 e0.1T j 3T z 1
1 2 e0.1T +j 3T z 1
further simplication using trigonometric relations yields H (z ) = 1 (e0.1T cos3T )z 1 1 2e0.1T cos3T z 1 + e0.2T Z 2
102
The magnitude of the frequency response of this lter is clearly associated with sampling interval T .
Example: To illustrate the design of lter using impulse invariance method transformation implied by (4.69), the specication of desired lter is illustrated in the gure below
|H (e j )| 1 1 1
p s
|H ( j )| 1 1 1
2
p T s T T
103
Bilinear transforms
Algebraic transformation between variables s and z , mapping of entire j axis of s- plane to one revolution of unit circle in the z plane. Denition: s= T denotes the sampling interval. The transfer function of the corresponding digital lter can be obtained from the transfer function of the analog lter Ha (s) according to H (z ) = H a Properties: solving (4.70) for z yields z= substituting s = + j we obtain z= 1 + T /2 + jT /2 1 T /2 jT /2 (4.73) 1 + (T /2)s 1 (T /2)s (4.72) 2 T 1 z 1 1 + z 1 (4.71) 2 T 1 z 1 1 + z 1 (4.70)
< 0 |z | < 1, > 0 |z | > 1 for all causal stable continuous time lters map in causal stable discrete-time lters. By inserting s = j in (4.72) it can be seen that |z | = 1 for all values of s on the j
axis. j -axis maps onto the unit circle.
104
Splane
j
Zplane
m 1111111111 0000000000 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111 0000000000 1111111111
1111 0000 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111 0000 1111
1 0
Relationship between and can be obtained from (4.70) with s = j and z = ej 2 j = T 2 = T 1 ej 1 + ej jsin(/2) cos(/2)
2 tan(/2) T
Nonlinear mapping between and = 2 tan(/2), T = 2tan1 (T /2) (4.74)
Design of a digital lter begins with frequency specication, which are converted into analog domain via (4.74). The analog lter is then designed and converted back into digital domain using the bilinear transforms.
= 2arctan( 2Td )
105
Example: Design a single pole lowpass digital lter with a 3-db bandwidth of 0.2 , using the bilinear transformation applied to an analog lter. H (s ) = c s + c
Where c is the 3-dB bandwidth at c = 0.2 . Solution:The digital lter is specied to have a 3-dB gain at c = 0.2 . In the analog frequency domain it corresponds to c = 2 tan(0.2/2) T 0.65 = T
Thus the analog lter has the system function H (s ) = 0.65/T s + 0.65/T
Inserting the denition of Bilinear from equation (4.70) we convert the analog lter in to digital lter H (z ) = 0.245(1 + z 1 ) 1 0.509z 1
The frequency response if this digital lter can be obtained as H () = 0.245(1 + ej ) 1 0.509z j
Please note that T has been divided out.
Chapter 5 Multirate signal processing

to be completed yet
5.1 5.2 5.3
Sampling rate reduction Sampling rate enhancement Filter banks

and N and N and s s N < s /2
P (j ) Vc (j )
N < c > ( s N )
106
Bibliography
[1] L.R. Bahl, J. Cocke, F. Jelinek, J. Raviv, Optimal Decoding of Linear Codes for Minimizing Symbol Error Rate, IEEE Transactions on Information Theory, Marz 1974 [2] J. Hagenauer, E. Oer, L. Papke, Iterative Decoding of Binary Block and Convolutional Codes, IEEE Transactions on Information Theory, Marz 1996 [3] M. Bossert, Kanalcodierung, 2. Auage 1998, B.G. Teubner Stuttgart [4] I. Land, Maximum a-posteriori Symbol-by-Symbol Estimation: LogMAP, Correspondence, April 2000 [5] I. Land, P. Hoher, On the Interpretation of the APP Algorithm as an LLR Filter, Proc. ISIT 2000, Sorrento, Italien, Juni 2000 ProbMAP vs.
107

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Air University Islamabad Department of Telecommunication Engineering. Prof. Dr.

Digital Signal Processing

Lecturer Asim Ali Samejo

1 Introduction 1.1 1.2 Signals,systems and signal processing . . . . . . . . . . . . . . . . . . . . . DSP applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3 Transform domains 3.1

4 Digital lters 4.1

Design of FIR lters . . . . . . . . . . . . . . . . . . . . . . . . . . 95 Design of IIR lters . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

Analog to digital transformation methods . . . . . . . . . . . . . . . . . . . 100

5 Multirate signal processing 5.1 5.2 5.3

Signals,systems and signal processing

What does Digital signal processing mean? Signal:

Passing the signal through a system. Examples:

System: Denition also includes:

Digital generally unlimited cost, complexity precision

Aging production costs Linear-phase frequency response Complex Algorithms

Without problems Low exactly realizable

Chapter 2 Classication of systems and signals

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

Common test signals

Unit impulse Unit sample is denoted by 0 (k ) 0 (k ) = 1, k = 0 0, elsewhere (2.1)

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

extension of unit impulse i.e.

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Rectangular pulses

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

T {av1 [k ] + bv2 [k ]} = aT {v1 [k ]} + bT {v2 [k ]}

Obviously y3 (k ) 0 k k if 1 s{v1 (k )} + 2 s{v2 (k )} = s{1 v1 (k ) + 2 v2 (k )} Example:

Digital processing of continuous signals

Generation of discrete-time signals from the continuous-time signals

we nally have the Fourier transform of vi (t) 1 Vi (j ) = T

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

) hr (t) = sinc(s t/2) c

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Combining (2.13),(2.20) in (2.19) yields

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

v (kT ) = v (k ) vn where vn is taken from a nite alphabet I .

instantenous amplitude Range of quantizer

Twos Complement binary offset

111 110 101 100 011 010 001 000

000 111 110 101 100

Transfer characteristic for midtreat quantizer

Quantization error eq (k ) with respect to the unquantized signal < eq (k ) 2 2

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

Quantization characteristic function for a midtreat quantizer with L = 8.

number representation has no inuence on the performance of quantization process.

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

Modeling of quantization noise

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS Zero mean, and a variance of

Inserting the denition of in (2.26) yields,

using (2.25) we obtain SNR = 10 log10

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

Nbit shift register

Serial to parallel A/D converters

N/2bit DAC N/2 bit

N/2bit ADC N/2 bit LSB

N/2bit DAC N/2 bit

N/2bit ADC N/2 bit LSB

MSB Ripple AD converter

Sample and Hold

LSB Flash Encoder

CHAPTER 2. CLASSIFICATION OF SYSTEMS AND SIGNALS

Characteristics of A/D converter

100 011 010 001 000 analog input