EE513 Audio Signals and Systems Complex Oscillator Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.

Slides:



Advertisements
Similar presentations
Signal Processing in the Discrete Time Domain Microprocessor Applications (MEE4033) Sogang University Department of Mechanical Engineering.
Advertisements

ECE 8443 – Pattern Recognition EE 3512 – Signals: Continuous and Discrete Objectives: Response to a Sinusoidal Input Frequency Analysis of an RC Circuit.
Copyright ©2010, ©1999, ©1989 by Pearson Education, Inc. All rights reserved. Discrete-Time Signal Processing, Third Edition Alan V. Oppenheim Ronald W.
Fundamentals of Electric Circuits Chapter 14 Copyright © The McGraw-Hill Companies, Inc. Permission required for reproduction or display.
Lecture 7: Basis Functions & Fourier Series
ECE 8443 – Pattern Recognition EE 3512 – Signals: Continuous and Discrete Objectives: Filters and Difference Equations Signal Flow Graphs FIR and IIR Filters.
EE513 Audio Signals and Systems Digital Signal Processing (Synthesis) Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
Digital Signal Processing – Chapter 11 Introduction to the Design of Discrete Filters Prof. Yasser Mostafa Kadah
CHE 185 – PROCESS CONTROL AND DYNAMICS
AMI 4622 Digital Signal Processing
MM3FC Mathematical Modeling 3 LECTURE 3
Lecture 19: Discrete-Time Transfer Functions
Implementation of Basic Digital Filter Structures R.C. Maher ECEN4002/5002 DSP Laboratory Spring 2003.
Frequency Response of Discrete-time LTI Systems Prof. Siripong Potisuk.
EE513 Audio Signals and Systems Wiener Inverse Filter Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
Introduction to Block Diagrams
Digital Signals and Systems
Lecture 9: Structure for Discrete-Time System XILIANG LUO 2014/11 1.
Unit III FIR Filter Design
EE513 Audio Signals and Systems Digital Signal Processing (Systems) Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
Discrete-Time and System (A Review)
Properties and the Inverse of
Automatic Control Theory-
DISCRETE-TIME SIGNALS and SYSTEMS
Chapter 6 Digital Filter Structures
Professor A G Constantinides 1 Signal Flow Graphs Linear Time Invariant Discrete Time Systems can be made up from the elements { Storage, Scaling, Summation.
EE Audio Signals and Systems Digital Signal Processing (Synthesis) Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
Copyright © 2001, S. K. Mitra Tunable IIR Digital Filters We have described earlier two 1st-order and two 2nd-order IIR digital transfer functions with.
Copyright © 2001, S. K. Mitra Digital Filter Structures The convolution sum description of an LTI discrete-time system be used, can in principle, to implement.
Prof. Brian L. Evans Dept. of Electrical and Computer Engineering The University of Texas at Austin EE 313 Linear Systems and Signals Spring 2013 Continuous-Time.
Husheng Li, UTK-EECS, Fall  Study how to implement the LTI discrete-time systems.  We first present the block diagram and signal flow graph. 
DISP 2003 Lecture 6 – Part 2 Digital Filters 4 Coefficient quantization Zero input limit cycle How about using float? Philippe Baudrenghien, AB-RF.
System Function of discrete-time systems
1 Z-Transform. CHAPTER 5 School of Electrical System Engineering, UniMAP School of Electrical System Engineering, UniMAP NORSHAFINASH BT SAUDIN
Course Outline (Tentative) Fundamental Concepts of Signals and Systems Signals Systems Linear Time-Invariant (LTI) Systems Convolution integral and sum.
EE513 Audio Signals and Systems
ELECTRICA L ENGINEERING Principles and Applications SECOND EDITION ALLAN R. HAMBLEY ©2002 Prentice-Hall, Inc. Chapter 6 Frequency Response, Bode Plots,
EE422 Signals and Systems Laboratory Infinite Impulse Response (IIR) filters Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
Dan Ellis 1 ELEN E4810: Digital Signal Processing Topic 7: Filter types and structures 1.Some filter types 2.Minimum and maximum phase 3.Filter.
Digital Filter Realization
Course Outline (Tentative) Fundamental Concepts of Signals and Systems Signals Systems Linear Time-Invariant (LTI) Systems Convolution integral and sum.
Digital Filter Structures
Fast Fourier Transforms. 2 Discrete Fourier Transform The DFT pair was given as Baseline for computational complexity: –Each DFT coefficient requires.
Structures for Discrete-Time Systems 主講人:虞台文. Content Introduction Block Diagram Representation Signal Flow Graph Basic Structure for IIR Systems Transposed.
Lecture 4: The z-Transform 1. The z-transform The z-transform is used in sampled data systems just as the Laplace transform is used in continuous-time.
ELECTRICAL ENGINEERING: PRINCIPLES AND APPLICATIONS, Third Edition, by Allan R. Hambley, ©2005 Pearson Education, Inc. CHAPTER 6 Frequency Response, Bode.
DISP 2003 Lecture 5 – Part 1 Digital Filters 1 Frequency Response Difference Equations FIR versus IIR FIR Filters Properties and Design Philippe Baudrenghien,
By: John Ernsberger. PURPOSE  The purpose of this project is to design an equalizer with both user controlled and hard set gains.
1 Eeng 224 Chapter 14 Frequency Response Huseyin Bilgekul Eeng 224 Circuit Theory II Department of Electrical and Electronic Engineering Eastern Mediterranean.
بسم الله الرحمن الرحيم Digital Signal Processing Lecture 14 FFT-Radix-2 Decimation in Frequency And Radix -4 Algorithm University of Khartoum Department.
EE 445S Real-Time Digital Signal Processing Lab Fall 2013 Lab 3 IIR Filters Chao Jia Debarati Kundu Andrew Mark.
Digital Signal Processing Lecture 9 Review of LTI systems
Application of digital filter in engineering
EE611 Deterministic Systems Examples and Discrete Systems Descriptions Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
EE422G Signals and Systems Laboratory Fourier Series and the DFT Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
Linear Constant-Coefficient Difference Equations
Review of DSP.
CHAPTER 5 Z-Transform. EKT 230.
EEE4176 Applications of Digital Signal Processing
By: Mohammadreza Meidnai Urmia university, Urmia, Iran Fall 2014
Laplace and Z transforms
EE Audio Signals and Systems
Description and Analysis of Systems
ME2300 DIGITAL SIGNAL PROCESSING [Slide 6] IIR Filter Design BY DREAMCATCHER
Quick Review of LTI Systems
Research Methods in Acoustics Lecture 9: Laplace Transform and z-Transform Jonas Braasch.
UNIT V Linear Time Invariant Discrete-Time Systems
Chapter 5 DT System Analysis : Z Transform Basil Hamed
Zhongguo Liu Biomedical Engineering
Review of DSP.
Presentation transcript:

EE513 Audio Signals and Systems Complex Oscillator Kevin D. Donohue Electrical and Computer Engineering University of Kentucky

Oscillator Design Marginally Stable approach: Design a system by placing poles so that a marginally stable system results, which oscillates with a fundamental frequency of f 0 when excited by a unit impulse. Show that TF and difference equation of oscillator system are given by: where K scales the input to control the amplitude of oscillations, and  controls the frequency of oscillation f o, and sampling frequency f s by:

Oscillator Design Trig-Identity Approach: Design a system by selecting values of A and B in the trig-identity below so that y[n] can substitute out the cos(nT  0 ) function (T is sampling period) and result in a second order autoregressive difference equation. Show that difference equation of oscillator system is given by: Oscillator is initiated with non-zero initial conditions. For n=0, let

Analyze Design Consider Z transform of general second order system: Show that Z transform can be expressed as: Input term Initial condition term

Multiple Frequency Oscillator Excite a bank of oscillators (in parallel) tuned to different frequencies Each term represents a separate difference equation (second order system) where their outputs can be added together:

Multiple-Frequency Oscillator To obtain a direct form implementation for use with the filter function in Matlab, each parallel term must be combined to obtain a higher order, but single term transfer function The numerator and denominator coefficients can be used directly in the direct-form I or II implementation of a complex (multiple-frequency) oscillator. The left hand side represents a parallel implementation. See Matlab functions residuez, filter, fdatool, dfilt

Direct Form I Implementations Direct form I implementation of an IIR filter. The square blocks represent unit delays, the triangles represent multiplies, and the circles represent accumulators. The variable w[n] is an intermediate value output from the all-zero component and the input to the all-pole component of the filter, as suggested by the factorization in the equation. z -1  b0b0 b0b0 b1b1 b1b1 b M-1 bMbM bMbM   x[n]x[n] w[n]w[n] y[n]y[n] z -1    -a 1 1/a 0 -a 2 -a N- 1

Direct Form II Implementation Direct form II implementation of an IIR filter. Note: 1. The difference from direct form I is that the input and w[n] are associated with the all-pole component of the filter while the output and w[n] are associated with the all-zero part as suggested by the equation below. 2. The filter coefficients are the same for either direct form I or II implementations. x[n]x[n] w[n]w[n] y[n]y[n] bMbM bMbM b1b1 b1b1 b0b0 b0b0 z -1 -a N -a 1 G0G0 G0G0 1/a 0   

Cascade Implementation From a direct form implementation a cascaded series of second order filters can be also be derived for another implementation of second order systems. Note in this case the coefficient are no longer the same as for the direct form implementations. In Matlab see method convert and sos and filter for dfilt, and tf2sos, sos2tf.

Parallel Implementation A parallel bank of second order filters can be also be derived (obtained directly from the oscillator design procedures described in these notes). Note in this case the coefficient are not the same as in the direct form or cascade implementations. In Matlab see method convert and parallel for dfilt.