15 Oct 2009Comp30291 Section 21 UNIVERSITY of MANCHESTER School of Computer Science Comp30291: Digital Media Processing 2009-10 Section 2 Analogue filtering.

Slides:



Advertisements
Similar presentations
ECE 8443 – Pattern Recognition EE 3512 – Signals: Continuous and Discrete Objectives: Response to a Sinusoidal Input Frequency Analysis of an RC Circuit.
Advertisements

Lecture 18: Linear convolution of Sequences and Vectors Sections 2.2.3, 2.3.
Digital Signal Processing IIR Filter IIR Filter Design by Approximation of Derivatives Analogue filters having rational transfer function H(s) can be.
Signal and System IIR Filter Filbert H. Juwono
24 Nov'09Comp30291 DMP Section 51 University of Manchester School of Computer Science Comp : Digital Media Processing Section 5 z-transforms & IIR-type.
Lecture 4 Active Filter (Part I)
Frequency Response and Filter Design By Poles and Zeros Positioning Dr. Mohamed Bingabr University of Central Oklahoma Slides For Lathi’s Textbook Provided.
ELEN 5346/4304 DSP and Filter Design Fall Lecture 8: LTI filter types Instructor: Dr. Gleb V. Tcheslavski Contact:
Department of Electronics and Communication Engineering, Manipal Institute of Technology, Manipal, INDIA Filters.
MALVINO Electronic PRINCIPLES SIXTH EDITION.
Leo Lam © Signals and Systems EE235. Transformers Leo Lam ©
Filtering Filtering is one of the most widely used complex signal processing operations The system implementing this operation is called a filter A filter.
Digital Signal Processing – Chapter 11 Introduction to the Design of Discrete Filters Prof. Yasser Mostafa Kadah
Lecture 16: Continuous-Time Transfer Functions
EE-2027 SaS, L15 1/15 Lecture 15: Continuous-Time Transfer Functions 6 Transfer Function of Continuous-Time Systems (3 lectures): Transfer function, frequency.
Active Filters Conventional passive filters consist of LCR networks. Inductors are undesirable components: They are particularly non-ideal (lossy) They.
ACTIVE FILTER CIRCUITS. DISADVANTAGES OF PASSIVE FILTER CIRCUITS Passive filter circuits consisting of resistors, inductors, and capacitors are incapable.
SAMPLING & ALIASING. OVERVIEW Periodic sampling, the process of representing a continuous signal with a sequence of discrete data values, pervades the.
EE313 Linear Systems and Signals Fall 2010 Initial conversion of content to PowerPoint by Dr. Wade C. Schwartzkopf Prof. Brian L. Evans Dept. of Electrical.
Lecture 29 Review: Frequency response Frequency response examples Frequency response plots & signal spectra Filters Related educational materials: –Chapter.
EE513 Audio Signals and Systems Digital Signal Processing (Systems) Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
DSP. What is DSP? DSP: Digital Signal Processing---Using a digital process (e.g., a program running on a microprocessor) to modify a digital representation.
EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing.
Chapter 5 Frequency Domain Analysis of Systems. Consider the following CT LTI system: absolutely integrable,Assumption: the impulse response h(t) is absolutely.
Fourier Analysis of Systems Ch.5 Kamen and Heck. 5.1 Fourier Analysis of Continuous- Time Systems Consider a linear time-invariant continuous-time system.
Vibrationdata 1 Unit 19 Digital Filtering (plus some seismology)
Sept'05CS32911 CS3291 Sect 2: Review of analogue systems Example of an analogue system: Represent as “black box” R C y(t) x(t) y(t)
17 Nov'09Comp30291: Section 41 University of Manchester School of Computer Science Comp30291 Digital Media Processing Section 4 ‘Design of FIR.
Sampling Theorems. Periodic Sampling Most signals are continuous in time. Example: voice, music, images ADC and DAC is needed to convert from continuous-time.
Nov'04CS3291: Section 41 University of Manchester Department of Computer Science CS3291 Digital Signal Processing '04-'05 Section 4: ‘A design technique.
16 Oct'09Comp30291 Section 31 University of Manchester School of Computer Science Comp30291: Digital Media Processing Section 3 : Discrete-time.
CISE315 SaS, L171/16 Lecture 8: Basis Functions & Fourier Series 3. Basis functions: Concept of basis function. Fourier series representation of time functions.
Nov '04CS3291: Section 61 UNIVERSITY of MANCHESTER Department of Computer Science CS3291: Digital Signal Processing Section 6 IIR discrete time filter.
IIR Filter design (cf. Shenoi, 2006) The transfer function of the IIR filter is given by Its frequency responses are (where w is the normalized frequency.
Chapter 5 Frequency Domain Analysis of Systems. Consider the following CT LTI system: absolutely integrable,Assumption: the impulse response h(t) is absolutely.
Oct'04CS32911 CS3291 : Digital Signal Processing '04-05 Section 3 : Discrete-time LTI systems 3.1Introduction: A discrete time system takes discrete time.
21 Oct'08Comp30291: Section 41 University of Manchester School of Computer Science Comp30291 Digital Media Processing Section 4 ‘Design of FIR.
Nov '04CS3291: Section 61 Baseado no material da: UNIVERSITY of MANCHESTER Department of Computer Science CS3291: Digital Signal Processing Section 6 IIR.
ELECTRICA L ENGINEERING Principles and Applications SECOND EDITION ALLAN R. HAMBLEY ©2002 Prentice-Hall, Inc. Chapter 6 Frequency Response, Bode Plots,
1 Introduction to Digital Filters Filter: A filter is essentially a system or network that selectively changes the wave shape, amplitude/frequency and/or.
1 Conditions for Distortionless Transmission Transmission is said to be distortion less if the input and output have identical wave shapes within a multiplicative.
ES97H Biomedical Signal Processing
Ch. 8 Analysis of Continuous- Time Systems by Use of the Transfer Function.
Technological Educational Institute Of Crete Department Of Applied Informatics and Multimedia Neural Networks Laboratory Slide 1 FOURIER TRANSFORMATION.
Hossein Sameti Department of Computer Engineering Sharif University of Technology.
Lecture 2: Filters.
Chapter 7. Filter Design Techniques
19 Nov'08Comp30291 Sectn 61 UNIVERSITY of MANCHESTER School of Computer Science Comp30291 : Digital Media Processing Section 6 Sampling & Reconstruction.
1 Prof. Nizamettin AYDIN Digital Signal Processing.
Digital Signal Processing
Filtering x y.
ELEC 202 Circuit Analysis II
Nov '03csDSP61 CS3291: Section 6 IIR discrete time filter design Introduction: Many design techniques for IIR discrete time filters have adopted ideas.
ELECTRICAL ENGINEERING: PRINCIPLES AND APPLICATIONS, Third Edition, by Allan R. Hambley, ©2005 Pearson Education, Inc. CHAPTER 6 Frequency Response, Bode.
Nov'04CS3291 Sectn 71 UNIVERSITY of MANCHESTER Department of Computer Science CS3291 : Digital Signal Processing ‘05 Section 7 Sampling & Reconstruction.
Digital Signal Processing Lecture 6 Frequency Selective Filters
Eeng360 1 Chapter 2 Linear Systems Topics:  Review of Linear Systems Linear Time-Invariant Systems Impulse Response Transfer Functions Distortionless.
Finite Impulse Response Filtering EMU-E&E Engineering Erhan A. Ince Dec 2015.
Electronics Technology Fundamentals Chapter 15 Frequency Response and Passive Filters.
Prof. Brian L. Evans Dept. of Electrical and Computer Engineering The University of Texas at Austin Lecture 3
Chapter 5 Active Filter By En. Rosemizi Bin Abd Rahim EMT212 – Analog Electronic II.
Digital Signal Processing
SAMPLING & ALIASING.
Lecture 14 Digital Filtering of Analog Signals
University of Manchester School of Computer Science
Analogue filtering UNIVERSITY of MANCHESTER School of Computer Science
Fourier Transform Analysis of Signals and Systems
HKN ECE 210 Exam 3 Review Session
Tania Stathaki 811b LTI Discrete-Time Systems in Transform Domain Ideal Filters Zero Phase Transfer Functions Linear Phase Transfer.
IIR Digital Filter Design
Presentation transcript:

15 Oct 2009Comp30291 Section 21 UNIVERSITY of MANCHESTER School of Computer Science Comp30291: Digital Media Processing Section 2 Analogue filtering

15 Oct 2009Comp30291 Section 22 Analog system represented as ‘black box’ x(t)y(t) Inside we could have analogue components, or Analog lowpass filter 1 ADC Digital processor DAC Analog lowpass filter 2 x(t) y(t) Fs

15 Oct 2009Comp30291 Section 23 Analog low-pass filters Analog Lowpass Filter 1 is ‘antialiasing’ filter: removes any frequency components above Fs/2 before sampling process. Analog Lowpass Filter 2 is ‘reconstruction’ filter: smoothes DAC output to remove all frequency components above Fs/2. Digital processor controls ADC to sample at Fs Hz. Also sends output sample to DAC at Fs samples per second. DAC produces ‘staircase’ waveform: smoothed by ALpF2. DAC output t

15 Oct 2009Comp30291 Section 24 Analogue filters Still needed in the world of DSP Also, many digital filter designs are based on analog filters. They are ‘linear’ & ‘time-invariant’ (LTI) Analogue filter x(t) y(t)

15 Oct 2009Comp30291 Section 25 System is LINEAR if (1)for any signal x(t), if x(t)  y(t) then a.x(t )  a.y(t) for any constant a. (2) for any signals x 1 (t) & x 2 (t), if x 1 (t)  y 1 (t) & x 2 (t)  y 2 (t) then x 1 (t) + x 2 (t)  y 1 (t) + y 2 (t) (By x(t)  y(t) we mean that applying x(t) to the input produces the output signal y(t). ) Definition of ‘linearity’

15 Oct 2009Comp30291 Section 26 Alternative definition of ‘linearity’ System is linear if for any signals x 1 (t) & x 2 (t), if x 1 (t)  y 1 (t) & x 2 (t)  y 2 (t) then a 1 x 1 (t) + a 2 x 2 (t)  a 1 y 1 (t) +a 2 y 2 (t) for any a 1 & a 2

15 Oct 2009Comp30291 Section 27 Linearity (illustration) Linear system If x 1 (t)  y 1 (t) & x 2 (t)  y 2 (t) then 3x 1 (t)+4x 2 (t)  3y 1 (t)+4y 2 (t) t x 2 (t) t y 2 (t) x 1 (t) t t y 1 (t) + +

15 Oct 2009Comp30291 Section 28 Definition of ‘time-invariance’ A time-invariant system must satisfy: For any x(t), if x(t)  y(t) then x(t-  )  y 1 (t-  ) for any  Delaying input by  seconds delays output by  seconds Not all systems have this property. An LTI system is linear & time invariant. An analogue filter is LTI.

15 Oct 2009Comp30291 Section 29 a 0 + a 1 s + a 2 s a N s N H(s) =  b 0 + b 1 s + b 2 s b M s M ‘System function’ for analogue LTI circuits An analog LTI system has a system (or transfer) function Coeffs a 0, a 1,...,a N, b 0,..., b M determine its behaviour. Designer of analog lowpass filters must choose these carefully. H(s) may be evaluated for complex values of s. Setting s = j  where  = 2  f gives a complex function of f. Modulus |H(j  )| is gain at  radians/second (  /2  Hz) Argument of H(j  ) is phase-lead at  radians/s.

15 Oct 2009Comp30291 Section 210 Gain & phase response graphs G(  Gain: G(  ) = |H(j  )| Phase lead:  (  ) = Arg[H(j  )| -()-() Gain Phase-lag  f / (2  )

15 Oct 2009Comp30291 Section 211 It may be shown that: when input x(t) = A cos(  t), output y(t) = A. G(  ). cos(  t +  (  ) ) Output is sinusoid of same frequency as input. ‘Sine-wave in  sine-wave out’ Multiplied in amplitude by G(  ) & ‘phase-shifted’ by  (  ). Example: If G(  ) = 3 and  (  ) =  /2 for all  what is the output? Answer: y(t) = 3.A.cos(  t +  /2) = 3.A.sin (  t) Effect of phase-response

15 Oct 2009Comp30291 Section 212 Express y(t) = A. G(  ). cos(  t +  (  ) ) as A. G(  ). cos (  [t +  (  )/  ]) = A. G(  ). cos(  [t -  (  )] ) where  (  ) = -  (  )/  Cosine wave is delayed by -  (  )/  seconds. -  (  )/  is ‘phase-delay’ in seconds Easier to understand than ‘phase-shift’ Phase-shift expressed as a delay

15 Oct 2009Comp30291 Section 213 If -  (  )/  is constant for all , all frequencies delayed by same time. Then system is ‘linear phase’ - this is good. Avoids changes in wave-shape due to ‘phase distortion’; i.e different frequencies being delayed by differently. Not all LTI systems are ‘linear phase’. Linear phase

15 Oct 2009Comp30291 Section 214 Linear phase response graph

15 Oct 2009Comp30291 Section 215 Low-pass analog filters Would like ideal ‘brick-wall’ gain response & linear phase response as shown below:   (  ) G(  ) 1 0 CC  C = cut-off frequency

15 Oct 2009Comp30291 Section 216 Butterworth low-pass gain response Cannot realise ideal ‘brick-wall’ gain response nor linear phase. Can realise Butterworth approximation of order n: Properties (i) G(0) = 1 ( 0 dB gain at  =0) (ii) G(  C ) = 1/(  2) ( -3dB gain at  =  C )

15 Oct 2009Comp30291 Section 217 Examples of Butterwth low-pass gain responses Let  C = 100 radians/second. G(  C ) is always 1/  (2) Shape gets closer to ideal ‘brick-wall’ response as n increases.

15 Oct 2009Comp30291 Section radians/second G(  ) n = 2 n=4 n=7 1 /  (2) LINEAR-LINEAR PLOT

15 Oct 2009Comp30291 Section 219 Butterworth gain responses on dB scale Plot G(  ) in dB, i.e. 20 log 10 (G(  )), against . With  on linear or log scale. As 20 log 10 (1/  (2)) = -3, all curves are -3dB when  =  C

15 Oct 2009Comp30291 Section dB radians/second -3dB dB-LINEAR PLOT n=2 n=4 n=7

15 Oct 2009Comp30291 Section dB radians/second dB-LOG PLOT n=2 n=4  3 dB

15 Oct 2009Comp30291 Section 222 clear all; for w = 1 : 400 G2(w) = 1/sqrt(1+(w/100)^4); G4(w) = 1/sqrt(1+(w/100)^8) ; G7(w) = 1/sqrt(1+ (w/100)^14); end; plot([1:400],G2,'r',[1:400],G4,'b',[1:400],G7,'k'); grid on; DG2=20*log10(G2); DG4=20*log10(G4); DG7=20*log10(G7); plot([1:400],DG2,'r',[1:400],DG4,'b',[1:400],DG7,'k'); grid on; semilogx([1:990], DG2,'r', [1:990], DG4, 'b’); MATLAB program to plot these graphs

15 Oct 2009Comp30291 Section 223 ‘Cut-off’ rate Best seen on a dB-Log plot Cut-off rate is 20n dB per decade or 6n dB per octave at frequencies  much greater than  C. Decade is a multiplication of frequency by 10. Octave is a multiplication of frequency by 2. So for n=4, gain drops by 80 dB if frequency is multiplied by 10 or by 24 dB if frequency is doubled.

15 Oct 2009Comp30291 Section 224  G(  ) 1 CC Low-pass with  C = 1 Low-pass  G(  ) 1 1 radian/s Ideal Approximatn Filter types - low-pass

15 Oct 2009Comp30291 Section 225   G(  ) 1 G(  ) 1 CC LL UU High-pass Band-pass Filter types - high-pass & band-pass

15 Oct 2009Comp30291 Section 226  G(  ) 1 LL UU Filter types - band-stop

15 Oct 2009Comp30291 Section 227   G(  ) 1 LL UU Narrow-band (  U < 2  L ) Broad-band (  U > 2  L ) Two types of band-pass gain-responses LL UU G(  ) 1

15 Oct 2009Comp30291 Section 228   G(  ) 1 LL UU Narrow-band (  U < 2  L ) Broad-band (  U > 2  L ) Three types of ‘band-stop’ gain-responses LL UU G(  ) 1

15 Oct 2009Comp30291 Section 229   G(  ) 1 NN Notch All-pass Third type of ‘band-stop’ gain-response Yet another type of gain-response G(  ) 1

15 Oct 2009Comp30291 Section 230 Approximatns for high-pass, band-pass etc Fortunately these can be derived from the formula for a Butterworth LOW-PASS gain response. MATLAB does all the calculations.

15 Oct 2009Comp30291 Section 231 A filter which uses a Butterworth gain-response approximation of order n is an ‘nth order Butterworth type filter’. In addition to Butterworth we have other approximations Chebychev (types 1 & 2) Elliptical Bessel, etc. Other approximations

15 Oct 2009Comp30291 Section 232 Problems 1. An analog filter has: H(s) = 1 / (1 + s) Give its gain & phase responses & its phase delay at  = Use MATLAB to plot gain response of Butterwth type analog low-pass filter of order 4 with  C = 100 radians/second. Solution to (2):- for w = 1 : 400 G(w) = 1/sqrt(1+(w/100)^8) ; end; plot(G); grid on;

15 Oct 2009Comp30291 Section 233 Result obtained:-