The Approximation Problem

Similar documents
The Approximation Problem

The Approximation Problem

EE 508 Lecture 15. Filter Transformations. Lowpass to Bandpass Lowpass to Highpass Lowpass to Band-reject

Digital Control & Digital Filters. Lectures 21 & 22

Filters and Tuned Amplifiers

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

EE482: Digital Signal Processing Applications

Filter Analysis and Design

Butterworth Filter Properties

Design of IIR filters

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

DIGITAL SIGNAL PROCESSING. Chapter 6 IIR Filter Design

Chapter 7: Filter Design 7.1 Practical Filter Terminology

Electronic Circuits EE359A

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

Digital Signal Processing

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

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

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

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

Digital Signal Processing

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

ω 0 = 2π/T 0 is called the fundamental angular frequency and ω 2 = 2ω 0 is called the

2.161 Signal Processing: Continuous and Discrete Fall 2008

Electronic Circuits EE359A

Shifted-modified Chebyshev filters

LINEAR-PHASE FIR FILTERS DESIGN

Quadrature-Mirror Filter Bank

Multimedia Signals and Systems - Audio and Video. Signal, Image, Video Processing Review-Introduction, MP3 and MPEG2

MITOCW watch?v=jtj3v Rx7E

Lecture 3 - Design of Digital Filters

Coefficients of Recursive Linear Time-Invariant First-Order Low-Pass and High-Pass Filters (v0.1)

Notes on L (Optimal) Filters

Let H(z) = P(z)/Q(z) be the system function of a rational form. Let us represent both P(z) and Q(z) as polynomials of z (not z -1 )

Stability Condition in Terms of the Pole Locations

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

All-Pole Recursive Digital Filters Design Based on Ultraspherical Polynomials

Digital Signal Processing IIR Filter Design via Bilinear Transform

INF3440/INF4440. Design of digital filters

CMPT 889: Lecture 2 Sinusoids, Complex Exponentials, Spectrum Representation

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

PS403 - Digital Signal processing

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

w n = c k v n k (1.226) w n = c k v n k + d k w n k (1.227) Clearly non-recursive filters are a special case of recursive filters where M=0.

Signal Conditioning Issues:

Lecture 27 Frequency Response 2

2.161 Signal Processing: Continuous and Discrete Fall 2008

A system that is both linear and time-invariant is called linear time-invariant (LTI).

Lecture 9 Infinite Impulse Response Filters

Frequency Response DR. GYURCSEK ISTVÁN

On the Frequency-Domain Properties of Savitzky-Golay Filters

Sinusoids. Amplitude and Magnitude. Phase and Period. CMPT 889: Lecture 2 Sinusoids, Complex Exponentials, Spectrum Representation

Digital Signal Processing Lecture 8 - Filter Design - IIR

ESS Finite Impulse Response Filters and the Z-transform

Transform Representation of Signals

1 1.27z z 2. 1 z H 2

Digital Signal Processing

EE 508 Lecture 4. Filter Concepts/Terminology Basic Properties of Electrical Circuits

GATE EE Topic wise Questions SIGNALS & SYSTEMS

Enhanced Steiglitz-McBride Procedure for. Minimax IIR Digital Filters

RATIONAL TRANSFER FUNCTIONS WITH MINIMUM IMPULSE RESPONSE MOMENTS

Poles and Zeros in z-plane

Analog and Digital Filter Design

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

The basic structure of the L-channel QMF bank is shown below

Sensors - February Demystifying Analog Filter Design

Optimum Ordering and Pole-Zero Pairing. Optimum Ordering and Pole-Zero Pairing Consider the scaled cascade structure shown below

Analysis of Finite Wordlength Effects

ON FINITE-PRECISION EFFECTS IN LATTICE WAVE-DIGITAL FILTERS

UNIT-II Z-TRANSFORM. This expression is also called a one sided z-transform. This non causal sequence produces positive powers of z in X (z).

An Iir-Filter Example: A Butterworth Filter

Design of Stable IIR filters with prescribed flatness and approximately linear phase

Lecture 16: Filter Design: Impulse Invariance and Bilinear Transform

ECE-700 Review. Phil Schniter. January 5, x c (t)e jωt dt, x[n]z n, Denoting a transform pair by x[n] X(z), some useful properties are

Power series solutions for 2nd order linear ODE s (not necessarily with constant coefficients) a n z n. n=0

MINIMUM PEAK IMPULSE FIR FILTER DESIGN

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

2.161 Signal Processing: Continuous and Discrete Fall 2008

Methods for Synthesizing Very High Q Parametrically Well Behaved Two Pole Filters

192 Chapter 4: Microwave Network Analysis

Optimal Design of Real and Complex Minimum Phase Digital FIR Filters

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 )

Matched Second Order Digital Filters

3 Fourier Series Representation of Periodic Signals

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

Time and Spatial Series and Transforms

(Refer Slide Time: 01:30)

x n cos 2x dx. dx = nx n 1 and v = 1 2 sin(2x). Andreas Fring (City University London) AS1051 Lecture Autumn / 36

Additional material: Linear Differential Equations

How to manipulate Frequencies in Discrete-time Domain? Two Main Approaches

ζ (s) = s 1 s {u} [u] ζ (s) = s 0 u 1+sdu, {u} Note how the integral runs from 0 and not 1.

f(g(x)) g (x) dx = f(u) du.

Need for transformation?

Digital Signal Processing:

Sinusoidal Forcing of a First-Order Process. / τ

Mathematics Specialist Units 3 & 4 Program 2018

( ) Frequency Response Analysis. Sinusoidal Forcing of a First-Order Process. Chapter 13. ( ) sin ω () (

2.004 Dynamics and Control II Spring 2008

EE 230 Lecture 24. Waveform Generators. - Sinusoidal Oscillators

Digital Signal Processing

Transcription:

EE 508 Lecture 3 The Approximation Problem Classical Approximating Functions - Thompson and Bessel Approximations

Review from Last Time Elliptic Filters Can be thought of as an extension of the CC approach by adding complex-conjugate zeros on the imaginary axis to increase the sharpness of the slope at the band edge Im TE j TA Re Ω S Concept Actual effect of adding zeros

Review from Last Time Elliptic Filters Magnitude-Squared Elliptic Approximating Function H = + C E Rn Inverse mapping to T E (s) exists For n even, n zeros on imaginary axis Termed here full order For n odd, n- zeros on imaginary axis Equal ripple in both pass band and stop band Analytical expression for poles and zeros not available Often choose to have less than n or n- zeros on imaginary axis (No longer based upon CC rational fractions)

Review from Last Time TE j Elliptic Filters If of full-order, response completely characterized by {n,ε,a S.Ω S } Any 3 of these paramaters are independent A S Typically ε,ω S, and A S are fixed by specifications (i.e. must determine n) Ω S

Review from Last Time TE j Elliptic Filters Observations about Elliptic Filters A S Ω S Elliptic filters have steeper transitions than CC filters Elliptic filters do not roll off as quickly in stop band as CC or even BW Highest Pole-Q of elliptic filters is larger than that of CC filters For a given transition requirement, order of elliptic filter typically less than that of CC filter Cost of implementing elliptic filter is comparable to that of CC filter if orders are the same Cost of implementing a given filter requirement is often less with the elliptic filters Often need computer to obtain elliptic approximating functions though limited tables are available Some authors refer to elliptic filters as Cauer filters

Review from Last Time Thompson and Bessel Approximations - All-pole filters - Maximally linear phase at =0

Review from Last Time Consider T(j) Thompson and Bessel Approximations phase R IM Nj NR j +jn j T j = = D j D j +jd j N tan I j D tan I j T j NR j DR j IM Phase expressions are difficult to work with Will first consider group delay and frequency distortion

Review from Last Time Linear Phase Consider T(j) R IM Nj NR j +jn j T j = = D j D j +jd j N tan I j D tan I j T j NR j DR j IM Defn: A filter is said to have linear phase if the phase is given by the expression T j θ where θ is a constant that is independent of

Review from Last Time Preserving the waveshape A filter has no frequency distortion for a given input if the output wave shape is preserved (i.e. the output wave shape is a magnitude scaled and possibly time-shifted version of the input) Mathematically, no frequency distortion for V IN (t) if V OUT (t)=kv IN (t-t shift ) Could have frequency distortion for other inputs

Review from Last Time Example of No Frequency Distortion Input 0.5 S 0-0.5 0 0 0 30 40 50 3-4 3 Output No Frequency Distortion S 4 0 - - -3 0 0 0 30 40 50 3-4

Review from Last Time Example with Frequency Distortion Input 0.5 S 0-0.5 0 0 0 30 40 50 3-5 4 3 0 - - -3 Output Phase Distortion (No Amplitude Distortion) 0 0 0 30 40 50 S 4-4 -5 3

Preserving wave-shape in pass band A filter is said to have linear passband phase if the phase in the passband of the filter is given by the expression T j θ where θ is a constant that is independent of If a filter has linear passband phase in a flat passband, then the waveshape is preserved provided all spectral components of the input are in the passband and the output can be expressed as an amplitude scaled and time shifted version of the input by the expression V OUT (t)=kv IN (t-t shift )

Preserving wave-shape in pass band X IN (s) Ts X OUT (s) Example: Consider a linear network with transfer function T(s) Assume In the steady state Xin t = Asin t+θ + Asint+θ X t = A T j sin t+θ T j + A T j sin t+θ T j OUT Rewrite as: T j T j XOUT t = A T j sin t+ +θ + A T j sin t+ +θ T If and are in a flat passband, j T j = T j Can express as: T j T j XOUT t = T j Asin t+ +θ + Asin t+ +θ

Preserving wave-shape in pass band X IN (s) Ts X OUT (s) Example: Xin t = Asin t+θ + Asint+θ If and are in a flat passband, T j = T j T j T j T j XOUT t = T j Asin t+ +θ + Asin t+ +θ If and are in a linear phase passband, T j = k and T j = k k k XOUT t = T j Asin t+ +θ + Asin t+ +θ X t = T j A sin t+k +θ + A sin t+k +θ OUT XOUT t = Tj xin t+k T j k

Preserving wave-shape in pass band X IN (s) Ts X OUT (s) Example: T j = T j XOUT t = Tj xin t+k Xin t = Asin t+θ + Asint+θ T j T j = k T j = k T j k This is a magnitude scaled and time shifted version ot the input so waveshape is preserved A weaker condition on the phase relationship will also preserve waveshape with two spectral components present T j T j

Amplitude (Magnitude) Distortion, Phase Distortion and Preserving wave-shape in pass band X IN (s) Ts X OUT (s) T j T j k If and are any two spectral components of an input signal in which T j T j then the filter exhibits amplitude distortion for this input. If and are any two spectral components of an input signal in which T j then the filter exhibits phase distortion for this input. T j If and are any two spectral components of an input signal that exhibits either amplitude or phase distortion for these inputs, then the waveshape will X t Hx t+k not be preserved OUT in

Amplitude (Magnitude) Distortion, Phase Distortion and Preserving wave-shape in pass band X IN (s) Ts X OUT (s) Amplitude and phase distortion are often of concern in filter applications requiring a flat passband and a flat zero-magnitude stop band Amplitude distortion is usually of little concern in the stopband of a filter Phase distortion is usually of little concern in the stopband of a filter A filter with no amplitude distortion or phase distortion in the passband and a zero-magnitude stop band will exhibit waveform distortion for any input that has a frequency component in the passband and another frequency component in the stopband It can be shown that the only way to avoid magnitude and phase distortion respectively for signals that have energy components in the interval << is to have constants k and k such that T j k T j k for < <

Group Delay Defn: Group Delay is the negative of the phase derivative with respect to G dt j d Recall, by definition, the phase is linear iff T j k d T j d k - k d d If the phase is linear, G Thus, the phase is linear iff the group delay is constant The group delay and the phase of a transfer function carry the same information But, of what use is the group delay?

Group Delay Example: Consider what is one of the simplest transfer functions Ts = s+ - T j= T j+ j = - tan The phase of T(s) is analytically very complicated d - -tan dt j G d Recall the identity - dtan u du dx u dx d G dt j d Thus - d -tan G d G Note that the group delay is a rational fraction in

But, of what use is the group delay? Group Delay The phase of almost all useful transfer functions are complicated functions involving Sums of arctan functions and these are difficult to work with analytically Theorem: The group delay of any transfer function is a rational fraction in From this theorem, it can be observed that the group delay is much more suited for analytical investigations than is the phase Proof of Theorem: (for notational convenience, will consider only all-pole transfer functions) T s = n k aks k=0

Group Delay Theorem: The group delay of any transfer function is a rational fraction in Proof of Theorem: Ts = n k aks k=0 T j = - a + a +... +j a - a + a +... 4 4 4 3 5 T j = F +jf where F and F are even polynomials in T j = - tan - F F

Group Delay Theorem: The group delay of any transfer function is a rational fraction in Proof of Theorem: T j = - tan - - F F but from identity d tan u du dx u dx F d d F G d F d + F Now consider the right-most term in the product d T j = - F F F F d d d d F d F F

Group Delay Theorem: The group delay of any transfer function is a rational fraction in Proof of Theorem: T j = - tan - F F Odd d F F F F d d d d F d F F Even Even Thus this term is an even rational fraction in

Group Delay Theorem: The group delay of any transfer function is a rational fraction in Proof of Theorem: G d F d F T j = - d F + F Even It follows that G is the product of rational fractions in so it is also a rational fraction in Although tedious, the results can be extended when there are zeros present in T(s) as well d

Thompson and Bessel Approximations - All-pole filters - Maximally linear phase at =0 since - All-pole filters - Maximally constant group delay at =0 - at =0 G G dt j d These criteria can be equivalently expressed as

Thompson and Bessel Approximations A T s = n k aks k=0 Must find the coefficients a 0, a, an to satisfy the constraints T j = - a + a +... +j a - a + a +... 4 4 4 3 5 Theorem: If T j = then G is given by the expression x + jy G dy dx x - y d d x + y This theorem is easy to prove using the identity given above, proof will not be given here

Thompson and Bessel Approximations A T s = n k aks k=0 Must find the coefficients a 0, a, an to satisfy the constraints T j = - a + a +... +j a - a + a +... 4 4 4 3 5 From this theorem, it follows that G a + a a -3a + 5a -3a a +a a +... 4 3 5 4 3 4 + a - a + a -a a +a +... 3 4 from the constraint at =0, it follows that a = G G To make maximally constant at =0, want to match as many coefficients in the numerator and denominator as possible starting with the lowest powers of from terms from 4 terms a a -3a a - a 3 5a -3a a +a a a -a a +a. 5 4 3 3 4

Thompson and Bessel Approximations A T s = n k aks k=0 Must find the coefficients a 0, a, an to satisfy the constraints It can be shown that the a k s are given by where k n-k! a for k n- an n-k H k! n-k! H n! H n n!

Thompson and Bessel Approximations A T s = n k aks k=0 Must find the coefficients a 0, a, an to satisfy the constraints Alternatively, if we define the recursive polynomial set by B s+... B = s + 3s + 3 B = k- B +s B k k- k- Then the n-th order Thompson approximation is given by Bn 0 TAn s = B s n Since the recursive set of polynomials are termed Bessel functions, this is often termed the Bessel approximation

Thompson and Bessel Approximations B 0 n TAn s = B n s http://www.rfcafe.com/references/electrical/bessel-poles.htm Poles of Bessel Filters lie on circle Circle does not go through the origin Poles not uniformly space on circumference

Thompson and Bessel Approximations Observations: B 0 n TAn s = B n s The Thompson approximation has relatively poor magnitude characteristic (at least if considered as an approximation to the standard lowpass function) The normalized Thompson approximation has a group delay of or a phase of at =0 Frequency scaling is used to denormalize the group delay or the phase to other values

Thompson and Bessel Approximations Use of Bessel Filters: X IN (s) X OUT (s) Ts Consider: -sh Ts = e -jh T j = e T j = cos-h +jsin-h T j = T j = -h If xin t XMsin t+θ x t X sint+θ-h x OUT OUT M M t X sin t-h +θ This is simply a delayed version of the input xout t xin t-h dt j = h xout t xin t- G (not realizable but can be approximated) But G d So, output is delayed version of input and the delay is the group delay

Thompson and Bessel Approximations Use of Bessel Filters: X IN (s) X OUT (s) Ts T s = e T j = -sh G T j = -h = h It is challenging to build filters with a constant delay A filter with a constant group delay and unity magnitude introduces a constant delay Bessel filters are filters that are used to approximate a constant delay Bessel filters are attractive for introducing constant delays in digital systems Some authors refer to Bessel filters as Delay Filters An ideal delay filter would - introduce a time-domain shift of a step input by the group delay - introduce a time-domain shift each spectral component by the group delay - introduce a time-domain shift of a square wave by the group delay

Thompson and Bessel Approximations Characterization of the step response of a filter From Introduction to the Theory and Design of Active Filters by Huelsman and Allen, p. 94-96

Thompson and Bessel Approximations Step Response of Butterworth Filter Delay is not constant Overshoot present and increases with order BW filters do not perform well as delay filters From Introduction to the Theory and Design of Active Filters by Huelsman and Allen, p. 94-96

Thompson and Bessel Approximations Step Response of Chebyschev Filter Delay is not constant Overshoot and ringing present and increases with order CC filters do not perform well as delay filters From Introduction to the Theory and Design of Active Filters by Huelsman and Allen, p. 94-96

Thompson and Bessel Approximations Step Response of Bessel Filters Delay becomes more constant as order increases No overshoot or ringing present Bessel filters widely used as delay filters Bessel filters often designed to achieve time-domain performance From Introduction to the Theory and Design of Active Filters by Huelsman and Allen, p. 94-96

Thompson and Bessel Approximations Comparison of Step Response of 3 rd -order Bessel, BW and CC filters From Continuous-Time Active Filter Design by Deliyannis, Sun and Fidler

Thompson and Bessel Approximations Harmonics in passband of Bessel Filter increase with n Attenuation of amplitude for Bessel does not compare favorably wth BW, CC, or Eliptic filters From Design of Analog Filters by Schaumann and Van Valkenburg, p405

Thompson and Bessel Approximations Magnitude of Bessel filters does not drop rapidly at band edge Phase of Bessel filters becomes very linear in passband as order increases From Continuous-Time Active Filter Design by Deliyannis, Sun and Fidler

Thompson and Bessel Approximations Comparison of Phase Response of 3 rd -order Bessel, BW and CC filters From Continuous-Time Active Filter Design by Deliyannis, Sun and Fidler

End of Lecture 3