Session 2 TD Filt

PEMP ESD2521
Session 2 : Review On Signals
Session delivered by: Chandan N.
M.S. Ramaiah School of Advanced Studies, Bengaluru
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Session Objectives
To understand basics operation on signals To understand the Time-Domain Characterization of LTI system To understand the effects of under sampling and over sampling To understand the concept of convloution To review on Time domain and Frequency domain signals
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Session Topics
Types of Signals Discrete time Systems Sampling Signal processing Aliasing
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Basic Sequences
1, n = 0 - [ n] = Unit sample sequence 1 0, n 0

n
4
3 2 1 0 1 2 3 4 5 6
Unit step sequence 1
1, n 0 [ n] = 0, n < 0
n
4
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Basic Sequences Real sinusoidal sequence x[n] = A cos(o n + ) where A is the amplitude, o is the angular frequency, and is the phase of x[n] Example = 0.1
o
2 1 Amplitude 0 -1 -2
10
20 Time index n
30
40
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Basic Sequences
n
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Exponential sequencex[ n] = A , < n <
, =e where A and are real or complex numbers If we write A = A e j , x[n] = then we can express xre [n] = A e
o n j ( o + jo ) n Ae e
( o + jo )
= xre [n] + j xim [n],
where o n xim [n] = A e sin(o n + )
cos(o n + ),
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Basic Sequences
1 0.5
xre [n]
Amplitude
0 -0.5 -1
and xim [n] of a complex exponential (o = 0) sequence are real sinusoidal sequences with(o > 0) constant (o < 0), growing and decaying Imaginary part amplitudes for n > 0 1
Real part 10
Amplitude
20 Time index n
30
40
0.5 0 -0.5 -1
10
20 Time index n
30
40
x[n] = exp( + j 6 )n
1 12
M.S. Ramaiah School of Advanced Studies, Bengaluru 7
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Basic Sequences
Real exponential sequence where A and a are real numbers
x[ n] = A n ,
< n <
= 1.2 50 40
= 0.9 20 15
Amplitude
20 10 0 0 5 10 20 15 Time index n 25 30
Amplitude
30
10 5 0 0
10
15 20 Time index n
25
30
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Basic Sequences
Sinusoidal sequence A cos(o n + ) and complex exponential sequence B exp( jo n) are periodic sequences of period N if o N = 2r where N and r are positive integers Smallest value of N satisfying o N = 2r is the fundamental period of the sequence To verify the above fact, consider
x1[n] = cos(o n + ) x2 [n] = cos(o (n + N ) + )

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Basic Sequences
Now
x2 [n] = cos(o (n + N ) + ) = cos(o n + ) cos o N sin(o n + ) sin o N cos o N = 1
which will be equal to only if and cos(o n + ) = x1[n]
sin o N = 0
These two conditions are met if and only if or
o N = 2 r
2 = N o r
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Basic Sequences
If 2 /o is a noninteger rational number, then the period will be a multiple of 2 /o Otherwise, the sequence is aperiodic Example - x[n] = sin( 3n + ) is an aperiodic sequence
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Basic Sequences
=0
0
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2 1.5
Amplitude
1 0.5 0 0
o = 0
10
20 Time index n
30
40
Here
2 r N= =1 0
Hence period
for r = 0
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Basic Sequences
= 0.1
0
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2 1
Amplitude
0 -1
o = 0.1
0.1
-2
10
20 Time index n
30
40
Here N = 2 r = 20 Hence for r = 1

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Basic Sequences An arbitrary sequence can be represented in the time-domain as a weighted sum of some basic sequence and its delayed (advanced) versions
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x[n] = 0.5[n+2] + 1.5[n-1] [n-2] + [n-4] + 0.75 [n-6]
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Discrete-Time Systems
A discrete-time system processes a given input sequence x[n] to generates an output sequence y[n] with more desirable properties In most applications, the discrete-time system is a single-input, single-output system:
Discrete time System
x[n] Input sequence
y[n] Output sequence
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Discrete-Time Systems: Examples
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2-input, 1-output discrete-time systems Modulator, adder 1-input, 1-output discrete-time systems Multiplier, unit delay, unit advance
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M-point moving-average system M 1 1 y[n] = x[n k ] M k =0
Used in smoothing random variations in data An application: Consider x[n] = s[n] + d[n], where s[n] is the signal corrupted by a noise d[n]
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s[n] = 2[n(0.9) n ], d[n] - random signal
8 6 Amplitude 4 2 0 -2 d[n] s[n] x[n]
10
20 30 Time index n
40
50
7 6 5 Amplitude 4 3 2 1 0 0 10 20 30 Time index n 40 50 s[n] y[n]
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Discrete-Time Systems Linear interpolation - Employed to estimate sample values between pairs of adjacent sample values of a discrete-time sequence Factor-of-4 interpolation
y[n]
3 0 1 2
4 5 6 7 8 9 10 11 12
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Discrete-Time Systems: Examples Factor-of-2 interpolator -
y[n] = xu [n] + 1 ( xu [n 1] + xu [n + 1]) 2 Factor-of-3 interpolator y[n] = xu [n] + 1 ( xu [n 1] + xu [n + 2]) 3 2 + ( xu [n 2] + xu [n + 1]) 3
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Impulse And Step Responses

The response of a discrete-time system to a unit sample sequence {d[n]} is called the unit sample response or simply, the impulse response, and is denoted by {h[n]} The response of a discrete-time system to a unit step sequence {m[n]} is called the unit step response or simply, the step response, and is denoted by {s[n]}
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Impulse Response Example - The impulse response of the system is obtained by setting x[n] = [n] resulting in
y[n] = 1x[n] + 2 x[n 1] + 3 x[n 2] + 4 x[n 3]
h[n] = 1 [n] + 2 [n 1] + 3 [n 2] + 4 [n 3]
The impulse response is thus a finite-length sequence of length 4 given by {h[n]} = {1, 2 , 3 , 4}
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Impulse Response
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Example - The impulse response of the discrete-time accumulator
y[n] =
x[]
is obtained by setting x[n] = [n] resulting in

h[n] =
=
[] = [n]
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Impulse Response
Example - The impulse response {h[n]} of the factor-of-2 interpolator 1 y[n] = xu [n] + ( xu [n 1] + xu [n + 1])
2
is obtained by setting xu [n] = [n]and is given by 1 h[n] = [n] + ( [n 1] + [n + 1])

2
The impulse response is thus a finite-length sequence of length 3: {h[n]} = {0.5, 1 0.5}
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Time-Domain Characterization of LTI Discrete-Time System

Input-Output Relationship - A consequence of the linear, time-invariance property is that an LTI discrete-time system is completely characterized by its impulse response Knowing the impulse response one can compute the output of the system for any arbitrary input
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Time-Domain Characterization of LTI Discrete-Time System Let h[n] denote the impulse response of a LTI discrete-time system We compute its output y[n] for the input:
x[n] = 0.5[n + 2] + 1.5[n 1] [n 2] + 0.75[n 5]
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As the system is linear, we can compute its outputs for each member of the input separately and add the individual outputs to determine y[n]
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Convolution Sum The summation

y[n] =
k =
x[k ] h[n k ] = x[n k ] h[n]

k =
is called the convolution sum of the sequences x[n] and h[n] and represented compactly as y[n] = x[n] * h[n]
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Convolution Sum
Properties Commutative property: x[n] * h[n] = h[n] * x[n] Associative property : (x[n] * h[n])* y[n] = x[n] * (h[n] * y[n]) Distributive property : x[n] * (h[n] + y[n]) = x[n] * h[n] + x[n] * y[n]
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Convolution Sum
Interpretation 1) Time-reverse h[k] to form h[k ] 2) Shift h[-k] to the right by n sampling periods if n > 0 or shift to the left by n sampling periods if h[n k ] n < 0 to form 3) Form the product v[k ] = x[k ]h[n k ] 4) Sum all samples of v[k] to develop the n-th sample of y[n] of the convolution sum
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Convolution Sum
Schematic Representation -
h[ k ]

h[n k ] v[k ] z
n
k
y[n]
The computation of an output sample using the convolution sum is simply a sum of products Involves fairly simple operations such as additions, multiplications, and delays
x[k ]
30

In practice, if either the input or the impulse response is of finite length, the convolution sum can be used to compute the output sample as it involves a finite sum of products If both the input sequence and the impulse response sequence are of finite length, the output sequence is also of finite length
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If both the input sequence and the impulse response sequence are of infinite length, convolution sum cannot be used to compute the output For systems characterized by an infinite impulse response sequence, an alternate time-domain description involving a finite sum of products will be considered
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Convolution Example Example - Develop the sequence y[n] generated by the convolution of the sequences x[n] and h[n] shown below
x[n]
3 1
0 1 2 3 4
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h[n]
2 1
3 0 1 2
n
1
1 2
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Convolution Example The sequence {y[n]} generated by the convolution sum is shown below
y[n]
5 3 1
2 3
0 2 1
1
4 5
1
7 6 8 9
2 3 4
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Simple Interconnection Schemes
Two simple interconnection schemes are:

Cascade Connection Parallel Connection
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Cascade Connection
h1[n] h2[n]
h2[n]
h1[n]
h1[n] * hh [n ] = h 2 [n ] [ n ]
h1[n]
h 2[n]
Impulse response h[n] of the cascade of two LTI discrete-time systems with impulse responses and is given by
h[n] = h1[n] * h 2[n]
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Cascade Connection
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The ordering of the systems in the cascade has no effect on the overall impulse response because of the commutative property of convolution A cascade connection of two stable systems is stable A cascade connection of two passive (lossless) systems is passive (lossless)
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Cascade Connection
An application is in the development of an inverse system If the cascade connection satisfies the relation
h1 [ n] h 2[n ] = [ n ]
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then the LTI system is said to be the inverse of and vice-versa
h1[n]
h 2[n]
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Parallel Connection
h1[n] h2[n]
h1[n] + hh [n ] = h 2 [n ] [ n ]
1
Impulse response h[n] of the parallel connection of two h1[n] h2[n] LTI discrete-time systems with impulse responses and is given by
h[n] = h1[n] + h2[n]

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Simple Interconnection Schemes Consider the discrete-time system where h1[n] = [n] + 0.5[n 1],
h2[n] = 0.5[n] 0.25[n 1],

h3[n] = 2[n],
h1[n]
n
+ +
h4[n] = 2(0.5) [n]
h2[n]
h3[n]
h4[n]
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Simple Interconnection Schemes Simplifying the block-diagram we obtain
h1[n] h2[n]
h3[ n ] + h 4[ n ]
h1[n]
h 2[ n ] * ( h3[ n ]+ h 4[ n ])
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Simple Interconnection Schemes Overall impulse response h[n] is given by

h[n] = h1[n] + h 2[n] * ( h3[n] + h 4[n]) = h1[n] + h 2[n] * h3[n] + h 2[n] * h 4[n]
Now, h2 [n] * h3[n] = ( 1 [n] 1 [n 1]) * 2[n]
= [n] 1 [n 1]
2
Finally
1 [ n 1] [ n] = [ n] h[n] = [n] + 1 [ n 1 ] + [ n ] 2 2
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Classification of LTI Discrete-Time Systems

Based on Impulse Response Length If the impulse response h[n] is of finite length, i.e.,
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h[n] = 0 for n < N1 and n > N 2 , N1 < N 2

then it is known as a finite impulse response (FIR) discretetime system The convolution sum description here is
y[n] =
N2
k = N1
h[k ]x[n k ]
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The output y[n] of an FIR LTI discrete-time system can be computed directly from the convolution sum as it is a finite sum of products Examples of FIR LTI discrete-time systems are the moving-average system and the linear interpolators
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If the impulse response is of infinite length, then it is known as an infinite impulse response (IIR) discrete-time system The class of IIR systems we are concerned with in this course are characterized by linear constant coefficient difference equations
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Classification of LTI Discrete-Time Systems Example - The discrete-time accumulator defined by y[n] = y[n 1] + x[n] is an IIR system
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Based on the Output Calculation Process Nonrecursive System - Here the output can be calculated sequentially, knowing only the present and past input samples Recursive System - Here the output computation involves past output samples in addition to the present and past input samples
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Based on the Coefficients Real Discrete-Time System - The impulse response samples are real valued Complex Discrete-Time System - The impulse response samples are complex valued
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Correlation of Signals
Definitions A measure of similarity between a pair of energy signals, x[n] and y[n], is given by the cross-correlation sequence rxy [] defined by
rxy [] =
n =
x[n] y[n ], = 0, 1, 2, ...
The parameter called lag, indicates the time-shift between the pair of signals
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There are applications where it is necessary to compare one reference signal with one or more signals to determine the similarity between the pair and to determine additional information based on the similarity For example, in digital communications, a set of data symbols are represented by a set of unique discrete-time sequences If one of these sequences has been transmitted, the receiver has to determine which particular sequence has been received by comparing the received signal with every member of possible sequences from the set Similarly, in radar and sonar applications, the received signal reflected from the target is a delayed version of the transmitted signal and by measuring the delay, one can determine the location of the target The detection problem gets more complicated in practice, as often the received signal is corrupted by additive ransom noise
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y[n] is said to be shifted by samples to the right with respect to the reference sequence x[n] for positive values of , and shifted by samples to the left for negative values of The ordering of the subscripts xy in the definition of rxy [] specifies that x[n] is the reference sequence which remains fixed in time while y[n] is being shifted with respect to x[n]
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If y[n] is made the reference signal and shift x[n] with respect to y[n], then the corresponding cross-correlation sequence is given by ryx [] = n = y[n]x[n ]
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Thus, ryx [] is obtained by time-reversing rxy [] = m = y[m + ]x[m] = rxy []
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Modern Communication System

Function View
Channel encoder Higher-layer Network protocols Channel decoder
Deinterleaver
Interelaver
Modulator
Tx filter Analog front end
Demodulator
Equalizer
Rx filter
Chip Architecture View

Synchronizer
PLL
SDRAM controller
Digital filters Equalizer
I-RAM D-RAM0 D-RAM1 D-cache I-cache
...
DSP core RISC core

peripheral bus PCMCIA bus
Viterbi decoder
Hardwired
signal processing / channel coding Digital Circuit
Logics for other peripheral
DMA controller
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Filtering Example Signals are usually a mix of useful information and noise How do we extract the useful information? Filtering is one way
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Filtering Example
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Filtering Equations
Let x[n] denote current input value (ECG + noise) x[n-1] is previous input value, x[n-k] is k-th previous input Let y[n] be the current filtered output value y[n-1] is previous output value , y[n-k] is k-th previous output Filtering operations carried out for this example: y[n] = 2.4*y[n-1] -2.6*y[n-2] + 1.5 y[n-3] 0.4*y[n-4] + 0.6*x[n] 1.9*x[n-1] + 2.8*x[n-2] -1.9*x[n-3] + 0.6*x[n-4]
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Transform Example
ki
Can you say which is 1 / # by looking at them? If not, go to frequency domain Another way to look at signals Done using transforms
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Transform Example
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Transform Equations Discrete Fourier Transform x Time domain signal X Frequency domain representation of x
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Correlation Equation Correlation x Transmitted signal y Received signal rxy- Correlation coefficients
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Amplification and Conditioning
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The voltage from a signal sensor is very small in magnitude. A microphone may produce voltages of the order of 10 -6 volts. Similarly for ECG sensors, vibration sensors etc. Prior to recording the signal or reproducing with an actuator an amplifier should signal condition by linearly amplifying the signal by an appropriate factor.
The above amplifier adds 60dB of gain (20log101000 = 60)
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Amplifier Distortion
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An amplifier which introduces unwanted artifacts, is said to be nonlinear and is, of course, very undesirable as it may mask signal components of interest.
The above amplifier is non-linear and actually outputs the input signal plus a 3rd order harmonic:
Unlike noise it is essentially impossible to remove the effects of distortion. Therefore we try to minimize the possibility of distortion by using suitable components.
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Signals and Noise
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Most acquired signals are corrupted by some level of noise which can cause information to be lost. Signal processing techniques are often used in an attempt to remove or attenuate noise. Most noise can be considered as additive (linear superposition) which can be address by linear filtering techniques.
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The Noise/Distortion Chain

Consider the various levels of noise and distortion added in a digital mobile communications link:
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DSP must minimize the amount of noise/distortion input to the chain, and where possible attenuate other sources.
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Signal to Noise Ratio
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Taking the logarithm of the linear signal power to noise power ratio (SNR) and multiplying by 10 gives the measure of decibels or dBs.
Recalling that Power is
Very low quality telephone line
Audio Cassette Deck

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Generic Analogue Signal Processing

In general an analogue signal processing system can be defined as a system that senses a signal to produce an analogue voltage, process this voltage, and reproduce the signal to its original analogue form. A public address system is an example of an analogue signal processing system:
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A Generic Input/Output DSP System
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A single input, single output DSP system has the following components:
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Generic Analogue Communications

For most base band telecommunications a voltage signal is transmitted over a cable.
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A simple example is a telephone. The acoustic signal is converted to a voltage which is then directly transmitted over a twisted pair of wires to be received at a remote location.
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Digital Data Communication
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Modern communications systems require that digital information is transmitted and received.
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Analogue to Digital Converter (ADC)
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An ADC is a device that can convert a voltage to a binary number, according to its specific input-output characteristic.
The number of digital samples converted per second is defined by the sampling rate of the converter, fs Hz.
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Digital to Analogue Converter (DAC)
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A DAC is a device that can convert binary numbers to voltages, according to its specific input-output characteristic.
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Signal Conditioning
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Note that prior to a signal being input to an ADC, an amplifier will be required to ensure that the full voltage range of the ADC is used this is referred to as signal conditioning. For the above ADC with a maximum input and output of 2 volts we would require that the input signal to the ADC has a similar range:
Depending on the output actuator, an amplifier, or at least a buffer amplifier will be required.
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Sampling
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The speed at which an ADC generates binary numbers is called the sampling rate or sampling frequency fs The time between samples is called the sampling period, ts:
Sampling frequency is quoted in samples per second, or simply as Hertz (Hz). The actual sampling rate will depend on parameters of the application. This may vary from: 10s of Hz for control systems, 100s of Hz for biomedical, 1000s of Hz for audio applications, 1,000,000s of Hz for digital radio front ends.
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Sampling an Analogue Signal
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After signal conditioning the ADC can produce binary number equivalents of the input voltage. If the ADC has finite precision due to a limited no. of discrete levels then there may be a small error associated with each sample.
The quantization step size is 0.0625 volts. If an 5 bit ADC is used, then the max/min voltage input is approx 0.0625 x 16 = 1 volt.
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Reproducing an Analogue Signal
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Using a DAC at an appropriate sampling rate, we can reproduce an analogue signal:
Note that the output is a little steppy caused by the zero order hold (step reconstruction); ....this artifact can however be removed with a reconstruction filter.
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First Order Hold

Alternatively a first order hold could be attempted in the DAC. Here the voltage between two discrete samples is approximated by a straight line.
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A first order apparently produces a more accurate reproduction of the analogue signal. However implementation of a circuit to perform interpolation is not trivial and turns out not to be necessary.
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Binary Data Word Lengths

Data word lengths for DSP applications, typically: Fixed Point Word lengths: Dynamic Range 8 bits 128 to +127 20 log 28 48 dB 16 bits 32768 to +32767 20 log 216 96 dB 24 bits 8388608 to +8388607 20log224 154 dB Floating Point Word lengths (for arithmetic only): 32 bits (1038 to +1038) (24 bit mantissa, 8 bit exponent)
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Note that data input from an ADC, or output to a DAC will always be fixed point, although the internal DSP computation may be floating point.
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Sampling How Fast ?
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To intuitively derive the sampling theorem, consider first a pure sine wave of frequency 100Hz:
In order to ensure that we retain all of the information in the signal what sampling rate should be used?
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Sampling Too Fast ?

Sampling at fs = 800Hz, i.e. 8 samples per period:
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Appears to be a reasonable sampling rate. Sampling at fs = 3000Hz, i.e. 30 samples per period:
Perhaps higher than necessary sampling rate

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Sampling Too Slow

Sampling at fs = 100Hz, i.e. 1 sample per period:
Signal interpreted as DC! Sampling at fs = 100Hz, i.e. 1 sample per period:
Most of the signal features are missed

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Suitable Sampling Rate
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From inspection of the above 100Hz digital waveforms at the four different sample rates: fs = 800Hz seems a reasonable sampling rate; fs = 3000Hz is perhaps higher than necessary; fs = 100Hz is too low and fails to correctly sample the waveform, and loses the signal parameter information; fs = 80Hz is too low and fails completely From mathematical theory the minimum sampling rate to retain all information is: greater than 2 x fmax where fmax is the maximum frequency component of a baseband, bandlimited signal.
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Signal Frequency Range Terminology

Nyquist frequency/rate: The Nyquist frequency, fn is identified as twice the maximum frequency component present in a signal. Baseband: The lowest signal frequency present is around 0 Hz:
fb = Bandwidth
Bandlimited: For all frequencies in the signal fh < f < fl fl = lowest freq fh = highest freq fb = fh fl
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Nyquist Sampling Rate
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If a baseband, band limited signal is composed of sine waves up to a frequency Hz, then
Nyquist frequency fn = 2fb
In we require to sample this signal and retain all information, then the sampling rate, must be chosen as: Fs > fn i.e. Fs > 2fb
This frequency is often referred to as the Nyquist sampling rate, (distinct from the Nyquist frequency).
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Aliasing
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When a (baseband) signal is sampled at a frequency below the Nyquist rate, then we lose the signal frequency information and aliasing is said to have occured. Aliasing can be illustrated by sampling a sine wave at below the Nyquist rate and then reconstructing. We note that it appears as a sine wave of a lower frequency (aliasing - cf. impersonating). Consider again sampling the 100Hz sine wave at 80Hz:
Reconstructed signal has a freq. of fs - fsignal = 20Hz

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Aliasing Example Consider the output from the following three systems:
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Aliased Spectra
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Anti Alias Filter
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Prior to the analogue to digital converter (ADC) all frequencies above fs/2 must be blocked or they will be interpreted as lower frequencies, i.e aliasing.
The anti-alias filter is analogue (ideally a brick wall filter), cutting off just before fs/2 Hz.
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Reconstruction Filter
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The analogue reconstruction filter at the output of a DAC removes the baseband image high frequencies present in the signal (in the form of the steps between the discrete levels).
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Zero Order Hold (ZOH)

Note that the operation of zero order hold can be interpreted as a simple reconstructing frequency filtering operation:
The step reconstruction therefore causes a droop near fs/2.

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Anti-Alias and Reconstruction
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Anti-alias and reconstructions filters are analogue, i.e. made from resistors, capacitors, amplifiers, even inductors. Ideally they are both very sharp cut off filters at frequency fs/2. In practice the roll off will be between 6dB/octave (a simple resistor and capacitor) to 96dB/octave (a 16th order filter).
Steeper roll-off is more expensive, but clearly for many applications, good analogue filters are essential. In a DSP system the accurately trimmed analogue filters could actually be more costly than the other DSP components: i.e. DSP processor, ADC, DAC, memory etc.
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Perfect Nyquist Sampling
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The Nyquist sampling theorem states that a (baseband) signal should sampled at greater than twice the maximum frequency component present in the signal: fs > 2 * fmax The sampled signal can then be perfectly reconstructed to the original analogue signal with no added noise or distortion.
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ADC Sampling Error
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Perfect signal reconstruction assumes that sampled data values are exact (i.e. infinite precision real numbers). In practice they are not, as an ADC will have a number of discrete levels. The ADC samples at the Nyquist rate, and the sampled data value is the closest (discrete) ADC level to the actual value:
Hence every sample has a small quantization error.

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ADC Sampling Error

Assume an ADC or quantizer has 5 bits of resolution and maximum/minimum voltage swing of +1 and -1 volts. The input/output characteristic is shown below:
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Quantization Error
If the smallest step size of a linear ADC is q volts, then the error of any one sample is at worst q/2 volts.
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Quantization Error
The quantization error is straightforward to calculate from: q = Vmax / 2N 1 where N is the number of bits in the converter. The dynamic range of an bit converter is often quoted in dBs: Dynamic Range=20log102N=20Nlog102=6.02N Therefore an 8 bit converter has a range of
ESD2521
Binary 10000000 to 01111111, or in decimal -128 to 127 has a dynamic range of approximately 48 dB.
95
PEMP
ADC Representation
The actual ADC can be represented by a sampler and a quantizer:
ESD2521
The quantization error of each sample is in the range and we can model the quantizer as a linear additive noise source.
96
PEMP
Timing Jitter Error
ESD2521
Note that when a signal is sampled there may be some jitter on the sampling clock which will cause additional sample error.
With jitter each sampling instant may be slightly offset, and therefore the sample value obtained and sent to the DSP will be in error.
PEMP ESD2521
Computation algorithms Fast algorithms

Computation-strength reduction of multiplications
Hardware-efficient algorithms
Arithmetic-strength reduction
Digit-level (bit-level) optimization Saving within processing units
Structure regularity
Finite precision problems

Truncation error propagation Limit cycle Overflow oscillation
98
PEMP ESD2521
Computation Algorithm Examples

LMS Algorithms
y (n) = k =0 x(n k ) wk (n)
M 1
e( n ) = d ( n ) y ( n )
wk (n + 1) = wk (n) + e(n) x(n k )

Some simplified algorithms
wk (n + 1) = wk (n) + sign (e(n)) x(n k ) wk (n + 1) = wk (n) + sign (e(n)) sign ( x(n k ))
wk (n + 1) = wk (n) + 2
+ log 2 (e ( n ) )
sign (e(n)) x(n k )

99
PEMP ESD2521
How is Signal Processed

Analog-Digital Conversion
waveform anti-alias LPF Bw = 1/(2Ts) sample & hold quantizer + coder digital data
latch
enable
pulse stream Ts : sampling period waveform analog LPF Amplitude mapper digital data
reg
Digital representation Digital signal processing
100
PEMP ESD2521
Sampling and Quantization

Sample and hold
3Ts 4Ts 5Ts 6Ts 7Ts 8Ts -4Ts-3Ts -2Ts -Ts 0 Ts
Quantization
10 9 8 7 6 5 4 3 2 1 -1 -4Ts-3Ts -2Ts -2 -3 -4 -5
3Ts 4Ts 5Ts 6Ts 7Ts 8Ts -Ts 0 Ts
101
PEMP ESD2521
Session Summary
Functional Architecture Hardware Architecture System-level algorithms vs Hardware computationefficient algorithms Trade-off and System performance and Hardware complexity (cost) Signal processing is focused on efficient implementation of integrated circuit. Signals can be classified as continuous-time and discrete time signals. A system is BIBO stable if its impulse response is absolutely summable. A response of an LTI system is the convolution of its impulse response and input

Session 2 TD Filt

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Session 2 TD Filt

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PEMP ESD2521

Session 2 : Review On Signals

Session delivered by: Chandan N.

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

1, n = 0 - [ n] = Unit sample sequence 1 0, n 0

Unit step sequence 1

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

Exponential sequencex[ n] = A , < n <

= xre [n] + j xim [n],

where o n xim [n] = A e sin(o n + )

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

x1[n] = cos(o n + ) x2 [n] = cos(o (n + N ) + )

x2 [n] = cos(o (n + N ) + ) = cos(o n + ) cos o N sin(o n + ) sin o N cos o N = 1

which will be equal to only if and cos(o n + ) = x1[n]

These two conditions are met if and only if or

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

Here N = 2 r = 20 Hence for r = 1

x[n] = 0.5[n+2] + 1.5[n-1] [n-2] + [n-4] + 0.75 [n-6]

M.S. Ramaiah School of Advanced Studies, Bengaluru

x[n] Input sequence

y[n] Output sequence

M.S. Ramaiah School of Advanced Studies, Bengaluru

Discrete-Time Systems: Examples

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

7 6 5 Amplitude 4 3 2 1 0 0 10 20 30 Time index n 40 50 s[n] y[n]

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

Discrete-Time Systems: Examples Factor-of-2 interpolator -

y[n] = xu [n] + 1 ( xu [n 1] + xu [n + 1]) 2 Factor-of-3 interpolator y[n] = xu [n] + 1 ( xu [n 1] + xu [n + 2]) 3 2 + ( xu [n 2] + xu [n + 1]) 3

M.S. Ramaiah School of Advanced Studies, Bengaluru

Impulse And Step Responses

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru 22

Example - The impulse response of the discrete-time accumulator

is obtained by setting x[n] = [n] resulting in

M.S. Ramaiah School of Advanced Studies, Bengaluru

is obtained by setting xu [n] = [n]and is given by 1 h[n] = [n] + ( [n 1] + [n + 1])

M.S. Ramaiah School of Advanced Studies, Bengaluru

Time-Domain Characterization of LTI Discrete-Time System

M.S. Ramaiah School of Advanced Studies, Bengaluru

Convolution Sum The summation

x[k ] h[n k ] = x[n k ] h[n]

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

Time-Domain Characterization of LTI Discrete-Time System

M.S. Ramaiah School of Advanced Studies, Bengaluru

Time-Domain Characterization of LTI Discrete-Time System

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru

Simple Interconnection Schemes

Two simple interconnection schemes are:

M.S. Ramaiah School of Advanced Studies, Bengaluru

h[n] = h1[n] * h 2[n]

M.S. Ramaiah School of Advanced Studies, Bengaluru

M.S. Ramaiah School of Advanced Studies, Bengaluru