Filters and Tuned Amplifiers

Similar documents
Electronic Circuits EE359A

Electronic Circuits EE359A

Speaker: Arthur Williams Chief Scientist Telebyte Inc. Thursday November 20 th 2008 INTRODUCTION TO ACTIVE AND PASSIVE ANALOG

OPERATIONAL AMPLIFIER APPLICATIONS

Homework Assignment 11

Exercise s = 1. cos 60 ± j sin 60 = 0.5 ± j 3/2. = s 2 + s + 1. (s + 1)(s 2 + s + 1) T(jω) = (1 + ω2 )(1 ω 2 ) 2 + ω 2 (1 + ω 2 )

Op-Amp Circuits: Part 3

Sophomore Physics Laboratory (PH005/105)

Digital Control & Digital Filters. Lectures 21 & 22

ECE 410 DIGITAL SIGNAL PROCESSING D. Munson University of Illinois Chapter 12

DIGITAL SIGNAL PROCESSING UNIT III INFINITE IMPULSE RESPONSE DIGITAL FILTERS. 3.6 Design of Digital Filter using Digital to Digital

Filter Analysis and Design

Digital Signal Processing

Electronic Circuits Summary

Single-Time-Constant (STC) Circuits This lecture is given as a background that will be needed to determine the frequency response of the amplifiers.

The Approximation Problem

(Refer Slide Time: 02:11 to 04:19)

Lectures: Lumped element filters (also applies to low frequency filters) Stub Filters Stepped Impedance Filters Coupled Line Filters

The Approximation Problem

EE482: Digital Signal Processing Applications

EE221 Circuits II. Chapter 14 Frequency Response

EE221 Circuits II. Chapter 14 Frequency Response

The Approximation Problem

PS403 - Digital Signal processing

( s) N( s) ( ) The transfer function will take the form. = s = 2. giving ωo = sqrt(1/lc) = 1E7 [rad/s] ω 01 := R 1. α 1 2 L 1.

UNIT - III PART A. 2. Mention any two techniques for digitizing the transfer function of an analog filter?

Digital Signal Processing Lecture 8 - Filter Design - IIR

First and Second Order Circuits. Claudio Talarico, Gonzaga University Spring 2015

Chapter 8: Converter Transfer Functions

EE 508 Lecture 11. The Approximation Problem. Classical Approximations the Chebyschev and Elliptic Approximations

Shifted-modified Chebyshev filters

Design of IIR filters

Some of the different forms of a signal, obtained by transformations, are shown in the figure. jwt e z. jwt z e

Analogue Filters Design and Simulation by Carsten Kristiansen Napier University. November 2004

Signals and Systems. Lecture 11 Wednesday 22 nd November 2017 DR TANIA STATHAKI

Case Study: Parallel Coupled- Line Combline Filter

Frequency response. Pavel Máša - XE31EO2. XE31EO2 Lecture11. Pavel Máša - XE31EO2 - Frequency response

R. W. Erickson. Department of Electrical, Computer, and Energy Engineering University of Colorado, Boulder

Today. 1/25/11 Physics 262 Lecture 2 Filters. Active Components and Filters. Homework. Lab 2 this week

-Digital Signal Processing- FIR Filter Design. Lecture May-16

DIGITAL SIGNAL PROCESSING. Chapter 6 IIR Filter Design

DC and AC Impedance of Reactive Elements

Active Filter Design by Carsten Kristiansen Napier University. November 2004

University of Illinois at Chicago Spring ECE 412 Introduction to Filter Synthesis Homework #4 Solutions

ECE 202 Fall 2013 Final Exam

Like bilateral Laplace transforms, ROC must be used to determine a unique inverse z-transform.

Sinusoidal Steady-State Analysis

Butterworth Filter Properties

Optimum Ordering and Pole-Zero Pairing of the Cascade Form IIR. Digital Filter

ECE3050 Assignment 7

Design of Narrow Band Filters Part 2

R-L-C Circuits and Resonant Circuits

Sensors - February Demystifying Analog Filter Design

192 Chapter 4: Microwave Network Analysis

ON THE USE OF GEGENBAUER PROTOTYPES IN THE SYNTHESIS OF WAVEGUIDE FILTERS

Frequency Response DR. GYURCSEK ISTVÁN

Time Series Analysis: 4. Linear filters. P. F. Góra

MITOCW watch?v=jtj3v Rx7E

Master Degree in Electronic Engineering. Analog and Telecommunication Electronics course Prof. Del Corso Dante A.Y Switched Capacitor

Analog and Digital Filter Design

Notes on L (Optimal) Filters

Lecture 4: R-L-C Circuits and Resonant Circuits

Electronics. Basics & Applications. group talk Daniel Biesinger

Dynamic circuits: Frequency domain analysis

Designing Information Devices and Systems II Fall 2018 Elad Alon and Miki Lustig Discussion 5A

Signal Conditioning Issues:

Cast of Characters. Some Symbols, Functions, and Variables Used in the Book

Electromagnetic Oscillations and Alternating Current. 1. Electromagnetic oscillations and LC circuit 2. Alternating Current 3.

8. Active Filters - 2. Electronic Circuits. Prof. Dr. Qiuting Huang Integrated Systems Laboratory

A Novel Tunable Dual-Band Bandstop Filter (DBBSF) Using BST Capacitors and Tuning Diode

Texas A&M University Department of Electrical and Computer Engineering

An example of answers for the final report of Electronics"

LECTURE 25 and MORE: Basic Ingredients of Butterworth Filtering. Objective: Design a second order low pass filter whose 3dB down point is f c,min

All-Pole Recursive Digital Filters Design Based on Ultraspherical Polynomials

Total No. of Questions :09] [Total No. of Pages : 03

Switched-Capacitor Circuits David Johns and Ken Martin University of Toronto

Second-order filters. EE 230 second-order filters 1

Laplace Transform Analysis of Signals and Systems

Driven RLC Circuits Challenge Problem Solutions

Solutions to Problems in Chapter 6

IIR digital filter design for low pass filter based on impulse invariance and bilinear transformation methods using butterworth analog filter

8.1.6 Quadratic pole response: resonance

Time Series Analysis: 4. Digital Linear Filters. P. F. Góra

Chapter 7: Filter Design 7.1 Practical Filter Terminology

2nd-order filters. EE 230 second-order filters 1

Solution: K m = R 1 = 10. From the original circuit, Z L1 = jωl 1 = j10 Ω. For the scaled circuit, L 1 = jk m ωl 1 = j10 10 = j100 Ω, Z L

MODULE-4 RESONANCE CIRCUITS

Schedule. ECEN 301 Discussion #20 Exam 2 Review 1. Lab Due date. Title Chapters HW Due date. Date Day Class No. 10 Nov Mon 20 Exam Review.

Appendix A Butterworth Filtering Transfer Function

DESIGN MICROELECTRONICS ELCT 703 (W17) LECTURE 3: OP-AMP CMOS CIRCUIT. Dr. Eman Azab Assistant Professor Office: C

Lowpass filter design for space applications in waveguide technology using alternative topologies

Lecture 16 FREQUENCY RESPONSE OF SIMPLE CIRCUITS

UNIVERSITY OF CALIFORNIA College of Engineering Department of Electrical Engineering and Computer Sciences

ECEN 326 Electronic Circuits

The general form for the transform function of a second order filter is that of a biquadratic (or biquad to the cool kids).

2.161 Signal Processing: Continuous and Discrete Fall 2008

H(s) = 2(s+10)(s+100) (s+1)(s+1000)

Impedance Matching and Tuning

Fourier Series Representation of

INFINITE-IMPULSE RESPONSE DIGITAL FILTERS Classical analog filters and their conversion to digital filters 4. THE BUTTERWORTH ANALOG FILTER

Transcription:

Filters and Tuned Amplifiers Essential building block in many systems, particularly in communication and instrumentation systems Typically implemented in one of three technologies: passive LC filters, active RC filters, switched-capacitor filters Passive LC filters work well at high frequencies, however, at low frequencies the required inductors are large, bulky and non-ideal. Furthermore, inductors are difficult to fabricate in monolithic form and are incompatible with many modern assembly systems Active RC filters utilize opamps together with resistors and capacitors and are fabricated using discrete, thick-film and thin-film technologies. The performance of these filters is limited by the performance of the opamps (e.g. frequency response, noise, offsets, etc.) Switched-capacitor filters are monolithic filters which typically offer the best performance in terms of cost. Fabricated using capacitors, switches and opamps. Generally poorer performance compared to passive LC or active RC filters

Butterworth For a Butterworth filter the optimum is flat response in the passband and steep slope soon after cutoff (maximally flat filter) The problem is similar to finding Fourier series of square wave. The solution lies in the use of Butterworth Polynomials 4 and 8 pole Butterworth filters common. 12 and 16 pole filters considered fairly sophisticated Standard table can be used to find component values Chebyshev In some applications the sharpness of the cutoff response is more important than the passband flatness By adding higher resonant peaks it is possible to obtain sharper cutoff at the expense of peaks in the passband

We also observe that the transmission decreases towards - as ω approaches. Thus the filter must have one more zero at s =. In general, the number of zeros at s = is the difference between the degree of the numerator polynomial, m, and the denominator polynomial, n This is because as s approaches, T(s) approaches a m / s n-m and thus is said to have n m zeros at s = For a filter circuit to be stable all its poles must lie in the left half of the s plane and thus all the poles must have negative real parts the figure on the next page shows typical pole and zero locations for the transmission function depicted in figure 11.3

We have assumed this filter is of fifth order (n = 5). It has two pairs of complex conjugate poles and one real axis pole All the poles lie in the vicinity of the passband which gives the filter its high transmission at passband frequencies The transfer function of the filter is of the form T ( s) = 5 ( 2)( 2 + ω ) L + ω 2 L2 a s s 4 2 1 s + b s + b s + b s + b s + b 4 4 3 As a further example consider the low pass filter below. We observe that there are no finite values of ω at which transmission is 0. Thus all zeros are at s =. The filter transfer function is of the form T( s) = am n s + b n 1 s + b n 2 s + b n 1 3 n 2 2 2 1 1 K 0 0

Such a filter is known as an all-pole filter

Butterworth Filter Butterworth and Chebyshev Filters The figure below shows the magnitude response of a Butterworth filter

The filter exhibits a monotonically decreasing transmission with all transmission zeros at ω =, making it an all pole filter The magnitude function for an n th order Butterworth filter with a passband edge ωp is given by T ( jω ) = 1+ ε 2 1 ω ω p 2n at ω = ω T(jω p ) p = 1 1+ ε 2 The parameter ε determines the maximum variation in passband transmission, A MAX (i.e. passband ripple) A MAX = 20log 1+ ε ( / 10) A MAX ε = 10 1 2

At the edge of the stopband ω = ωs A( ω ) = 20log 1 S = 10 log 1 + ε 2 ω ω 1+ ε S P 2 2n ω ω This equation can be used to determine the filter order required, which is the lowest value of n that yields A(ωS) A MIN The poles of a Butterworth filter can be determined from the graphical construction below The poles lie on a circle of radius ωp(1/ε) 1/n and are spaced at equal angles of π/n Since all poles have equal radial distance from the origin they all have the same frequency S P 2n ω ο = ω 1 P ε 1 n

Once the poles are known the transfer funciton can be written as T ( s) = n kω ο 1 2 K ( s p )( s p ) ( s p ) where k is a constant equal to the required DC gain of the filter n

The Chebyshev Filter The figure below shows representative transmission functions for Chebyshev filters of even and odd order The Chebyshev filter exhibits an equiripple response in the passband and a monotonically decreasing transmission in the stopband The total number of passband maxima and minima equals the order of the filter All the transmission zeros of the Chebyshev filter are at ω = making it an all pole filter The magnitude of the transfer function of an n th order Chebyshev filter with a passband edge ω P is given by T ( jω ) = 1 ω ω 2 2 + ε cos 1 1 n cos ω ω p p T ( jω ) = 1 ω ω 2 2 + ε cosh 1 1 ncosh ω ω p p

At the passband edge ω = ω P T ( jω ) = 1 1+ ε 2 The parameter ε determines the passband ripple according to A MAX ε = 10 log 1+ ( / 10) 2 ( ε ) Α X = ΜΑ 10 1 The attenuation achieved by the Chebyshev filter at the stopband edge (ω = ω S ) is A( ω S ) = 10 log 1 + ε cosh cosh ( ) ω n S ω 2 2 1 The poles of a Chebyshev filter are given by P k k = 2 1π p 1 1 1 ω sin sinh sinh n 2 n ε k + j 2 1π ω p 1 1 cos cosh sinh n 2 n ε P

The transfer function of the Chebyshev filter can be written as T ( s) = n k ω n p ( s p )( s p ) ( s p ) ε 2 1 1 2 K n The Chebyshev filter provides greater stopband attenuaton than the Butterworth filter at the expense of passband ripple

Bessel (Tompson) Response The sharper the filter cutoff the worse the phase shift right before cutoff (response after cutoff usually doesn t matter as the signal is attenuated) Poor phase response results in unequal delay which results in poor transient response. For a low pass filter this would normally result in output ringing when given a high frequency response Bessel (Tompson) filter is a maximally flat filter which provides linear phase shift. (i.e. equal delay). A Bessel filter can be used to compensate phase shifts introduced by other parts of the system

Filter Transfer Function The filter transfer function can be written as the ration of two polynomials T( s) = m m 1 ams + am 1s + K + a n n 1 s + b s + K + b n 1 The degree of the denominator, n, is the filter order. For the filter to be stable m n The numerator and denominator coefficients are real numbers. The polynomials in the numerator and denominator can be factored and T(s) can be expressed in the form T ( s) = am ( s z1)( s z2 ) K( s zm ) ( s p )( s p ) K( s p ) 1 2 where the z s are the zeroes and the p s are the poles Each zero or pole can be a real or complex number. Complex zeroes and poles, however, must occur in conjugate pairs. Thus if -1 + j2 is a zero then 1 + j2 is a zero o n o

Frequency Transformations We can transform a low pass filter to a high pass filter using the following transformation p = 1 s Third order high-pass Butterworth given by T ( s) = k s + s + = 3 1 2 1 2 2 1 s + 1 3 k s 3 2 s + 2 s + 2 s + 1 The high pass response will have the same cutoff frequency and passband flatness Usually there is a simple transformation of C s, R s and L s which will depend on the circuit topology

Low pass Band pass p = s ( ω 3dB ) ( ω 3dB ) s Low pass Band reject p = s 2 2 + ω ο s 2 2 + ω ο ω ο = Center Frequency Since in the filter stopband the transmission is required to be zero or small, the filter zeroes are usually placed on the jω axis at stopband frequencies

This particular filter can be seen to have infinite attenuation (zero transmission) at two stop band frequencies ωl1 and ωl2 the filter must have zeros at s = + jωl1 and s = + jωl2 however, since complex zeros occur in conjugate pairs there must also be zeros at s = jωl1 and s = jωl2 Thus the numerator polynomial of this filter will have the factors ( s+ jω )( s jω )( s+ jω )( s jω ) L1 L1 L2 L2 which can be written as ( 2 2 s )( 2 + ω s 2 ) L1 + ω L2 For s = jω the numerator becomes ( ω 2 + ω 2 )( ω 2 + ω 2 ) L1 L2 which is zero at ω = ωl1 and ω = ωl2

The section cutoff frequencies and Q factors can be determined using Chebyshev Polynomials. Whereas for the Butterworth filter we only specify the number of poles (zeroes), for a Chebyshev filter we specify the number of poles (zeroes) and passband flatness. (i.e. a 0.5 db Chebyshev filter has a maximum peak 0.5 db above the minimum valley in the passband (Equal - Ripple filter) Elliptical (Cauer) Response The filter cutoff response can be improved further by following a basic low pass filter with a notch filter. The notch decreases the response just after cutoff To be effective the notch has to be narrow and as a result the overall response will start to increase just after the notch To eliminate this we add a number of notches until the original filter curve has dropped low enough Filter is specified with three parameters: passband ripple, order, minimum stop band attenuation Due to the large number of options it is difficult to find tables

Filters are generally linear circuits that can be represented as a two port network

The filter transfer function is given as follows T ( jω ) = T ( s) V V o i ( s) ( s) The magnitude of transmission is often expressed in decibels in terms of the gain function G ( ω) = 20log T ( jω) db A filter shapes the frequency spectrum of the input signal, according to the magnitude of the transfer function The phase characteristics of the signal are also modified as it passes through the filter Filters can be classified into a number of categories based on which frequency bands are passed through and which frequency bands are stopped

The above transmission characteristics are for ideal filters. Realistic transmission characteristics for a low pass filter are shown below

Transmission of a low pass filter is specified by four parameters 1. Passband edge, ω P 2. Maximum allowed variation in passband transmission, A MAX 3. Stopband edge, ω S 4. Minimum required stopband attenuation, A MIN The ratio ω S / ω P is usually used to measure the sharpness of the filter response and is called the selectivity factor The more tightly one specifies a filter (i.e. lower A MAX, higher A MIN, ω P / ω S closer to unity) the resulting filter must be of higher order and thus more complex and expensive Common to refer to A MAX as the passband ripple The process of obtaining a transfer function that meets given specifications is known as filter approximation. Filter approximation is usually performed using computer programs or filter design tables. In simpler cases, filter approximation can be performed using closed form expressions

The figure below shows transmission specifications for a bandpass filter

General Notes Each capacitive / inductive element adds a pole or zero to the frequency response By adding additional poles / zeroes we can increase the rolloff of the filter response (e.g. two pole low-pass filter would have -40 db / decade rolloff at high frequency) In adding poles we must take into account the loading of successive stages

Addition of buffer removes impedance matching effects When capacitors are combined with inductors it is possible to make circuits with very sharp frequency response

T ( s) = = 1 sc 1 = 1 2 R+ sl+ s L C + src + 1 sc s ω ο ω ο + Q s + 2 2 ω ο ω ο = 1 LC Q L 1 = = R L C ω RC

Two pole rolloff (40 db / decade) Q of circuit affects response When Q = 1/2 response is equivalent to two pole RC filter High Q pumps up gain below cutoff and increases slope after cutoff Single RC filter has large passband ripple and shallow rolloff Addition of RC stages increases rolloff, however passband flatness not affected and using a large number of stages does not give sharp cutoff for the first 50 db. 8 section RLC is better than 32 section RC By adjusting the Q of each section of a filter it is possible to achieve better response For a two section filter we can increase the Q of the second stage so that the peak fills in the rounded area just beyond cutoff of the first section The scheme improves the passband flatness just short of cutoff and allows us to reach ultimate slope just after cutoff How do we select the Q to give optimum performance and what is our optimum?