EE513 Audio Signals and Systems Digital Signal Processing (Systems) 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

Lecture 7: Basis Functions & Fourier Series
Digital Signal Processing IIR Filter IIR Filter Design by Approximation of Derivatives Analogue filters having rational transfer function H(s) can be.
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.
Signal and System IIR Filter Filbert H. Juwono
Infinite Impulse Response (IIR) Filters
So far We have introduced the Z transform
Digital Signal Processing – Chapter 11 Introduction to the Design of Discrete Filters Prof. Yasser Mostafa Kadah
ELEN 5346/4304 DSP and Filter Design Fall Lecture 7: Z-transform Instructor: Dr. Gleb V. Tcheslavski Contact:
AMI 4622 Digital Signal Processing
IIR Filter Design: Basic Approaches Most common approach to IIR filter design: (1)Convert specifications for the digital filter into equivalent specifications.
Lecture 19: Discrete-Time Transfer Functions
EE-2027 SaS, L11 1/13 Lecture 11: Discrete Fourier Transform 4 Sampling Discrete-time systems (2 lectures): Sampling theorem, discrete Fourier transform.
EE-2027 SaS, L18 1/12 Lecture 18: Discrete-Time Transfer Functions 7 Transfer Function of a Discrete-Time Systems (2 lectures): Impulse sampler, Laplace.
Frequency Response of Discrete-time LTI Systems Prof. Siripong Potisuk.
Image (and Video) Coding and Processing Lecture 2: Basic Filtering Wade Trappe.
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.
Echivalarea sistemelor analogice cu sisteme digitale Prof.dr.ing. Ioan NAFORNITA.
Discrete-Time and System (A Review)
1 Chapter 8 The Discrete Fourier Transform 2 Introduction  In Chapters 2 and 3 we discussed the representation of sequences and LTI systems in terms.
UNIT - 4 ANALYSIS OF DISCRETE TIME SIGNALS. Sampling Frequency Harry Nyquist, working at Bell Labs developed what has become known as the Nyquist Sampling.
ECE 8443 – Pattern Recognition EE 3512 – Signals: Continuous and Discrete Objectives: Stability and the s-Plane Stability of an RC Circuit 1 st and 2 nd.
Properties of the z-Transform
CE Digital Signal Processing Fall 1992 Z Transform
University of Khartoum -Signals and Systems- Lecture 11
EE Audio Signals and Systems Digital Signal Processing (Synthesis) Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
UNIT I. SIGNAL ► Signal is a physical quantity that varies with respect to time, space or any other independent variable Eg x(t)= sin t. Eg x(t)= sin.
CHAPTER 4 Laplace Transform.
Fourier Series. Introduction Decompose a periodic input signal into primitive periodic components. A periodic sequence T2T3T t f(t)f(t)
CHAPTER 4 Laplace Transform.
Chapter 5 Z Transform. 2/45  Z transform –Representation, analysis, and design of discrete signal –Similar to Laplace transform –Conversion of digital.
EE313 Linear Systems and Signals Fall 2005 Initial conversion of content to PowerPoint by Dr. Wade C. Schwartzkopf Prof. Brian L. Evans Dept. of Electrical.
1 Z-Transform. CHAPTER 5 School of Electrical System Engineering, UniMAP School of Electrical System Engineering, UniMAP NORSHAFINASH BT SAUDIN
Department of Computer Eng. Sharif University of Technology Discrete-time signal processing Chapter 3: THE Z-TRANSFORM Content and Figures are from Discrete-Time.
1 Lecture 1: February 20, 2007 Topic: 1. Discrete-Time Signals and Systems.
Z TRANSFORM AND DFT Z Transform
EE313 Linear Systems and Signals Spring 2013 Initial conversion of content to PowerPoint by Dr. Wade C. Schwartzkopf Prof. Brian L. Evans Dept. of Electrical.
ES97H Biomedical Signal Processing
Technological Educational Institute Of Crete Department Of Applied Informatics and Multimedia Neural Networks Laboratory Slide 1 FOURIER TRANSFORMATION.
Digital Signal Processing
Signal and Systems Prof. H. Sameti Chapter 10: Introduction to the z-Transform Properties of the ROC of the z-Transform Inverse z-Transform Examples Properties.
Digital Signal Processing
Z Transform The z-transform of a digital signal x[n] is defined as:
Technological Educational Institute Of Crete Department Of Applied Informatics and Multimedia Neural Networks Laboratory Slide 1 DISCRETE SIGNALS AND SYSTEMS.
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 5 – 6 Z - Transform By Dileep Kumar.
Lecture 3: The Sampling Process and Aliasing 1. Introduction A digital or sampled-data control system operates on discrete- time rather than continuous-time.
Fourier Representation of Signals and LTI Systems.
Topics 1 Specific topics to be covered are: Discrete-time signals Z-transforms Sampling and reconstruction Aliasing and anti-aliasing filters Sampled-data.
2D Fourier Transform.
DISP 2003 Lecture 5 – Part 1 Digital Filters 1 Frequency Response Difference Equations FIR versus IIR FIR Filters Properties and Design Philippe Baudrenghien,
1 Fourier Representation of Signals and LTI Systems. CHAPTER 3 School of Computer and Communication Engineering, UniMAP Amir Razif B. Jamil Abdullah EKT.
Oh-Jin Kwon, EE dept., Sejong Univ., Seoul, Korea: 2.3 Fourier Transform: From Fourier Series to Fourier Transforms.
Digital Signal Processing Lecture 6 Frequency Selective Filters
Digital Signal Processing
CHAPTER 5 Z-Transform. EKT 230.
Speech Signal Processing
Sampling and Quantization
Laplace and Z transforms
EE Audio Signals and Systems
Chapter 8 Design of Infinite Impulse Response (IIR) Digital Filter
UNIT II Analysis of Continuous Time signal
Quick Review of LTI Systems
Research Methods in Acoustics Lecture 9: Laplace Transform and z-Transform Jonas Braasch.
Chapter 5 DT System Analysis : Z Transform Basil Hamed
Z TRANSFORM AND DFT Z Transform
Discrete-Time Signal processing Chapter 3 the Z-transform
Lecture 2: Signals Concepts & Properties
DIGITAL CONTROL SYSTEM WEEK 3 NUMERICAL APPROXIMATION
Presentation transcript:

EE513 Audio Signals and Systems Digital Signal Processing (Systems) Kevin D. Donohue Electrical and Computer Engineering University of Kentucky

Laplace to begin with The Laplace transform is used to characterize and analyze signal and system interactions. If x(t) is the time domain signal, its Laplace transform is defined as: where s belongs to the set of complex numbers over which the integral converges. What is the Laplace transform of a system impulse response? What does the Laplace transform become if it is evaluated along the imaginary axis?

Z-Transform If the Laplace transform is made discrete by sampling the time axis with interval T s, it becomes: Now let:

Z-Transform Substitute z to obtain: Normalize sampling rate and define Z-transform as: where z belongs to the set of complex numbers for which summation converges. Describe mapping from S-plane to Z- plane.

j  Axis in s to z Mapping j  axis in s corresponds to a zero real part,  = 0 for all imaginary values, . In z this results in: S-plane IM RE Z-plane IM

Z-plane S-plane Relationships Imaginary axis (j  ) in S-plane, maps into the unit circle in Z- plane, where segments on the j  axis described by: map into a complete unit circle for every integer k. Since the mapping from S to Z is not unique, let the range for k=0 be referred to as the root frequency range and all other values of k with the aliased frequency ranges.

Negative Real Axis s to z Mapping Negative real axis in s corresponds to  < 0. In z this results in: Z-plane RE IM S-plane IM RE

Z-plane S-plane Relationships The negative real axis of the S-plane (  < 0 ) maps into the area inside the unit circle of Z (| z | < 1). Therefore the stable region of the S-plane (left-half plane) corresponds to the area inside the unit circle of the Z-plane. Similarly, the unstable region of the S-plane (right-half plane) corresponds to the area outside the unit circle of the Z- plane. Because of aliasing from the sampling and scaling from the exponential transformation, there is no simple (linear) scaling between the Z and S planes. A warping or distortion occurs when matching domain points: Warping Possible Aliasing

Z-Transform One-Sided Many applications assume the input starts at t = 0 (n=0 for discrete) and no response exists before t = 0. So the Z-transform is often written as: Examples: Find the z-transforms of x[n] = u[n] and x[n] = a n u[n]; Assuming the z-transform of x[n] is X(z), find the Z-transform of x[n-k] for k>0.

Homework(1) Find the z-Transforms of: a) b) c) Use definition for k > 0. Use definition Use ZT properties (delay)

Convolution Given the impulse response of a discrete linear system h[n] the input-output relationship is described by discrete convolution: For x[n] and h=[n] below, graphically demonstrate their convolution.

Sinusoidal Response Consider sinusoidal input: Note this input is always on (steady-state). Show that for impulse response h(n), the convolution sum for evaluating the output becomes: Important Concepts:  The response to a sinusoidal input is a phasor multiplication between input phasor and transfer function value at the excitation frequency.  The frequency response (Transfer Function) of a discrete system is the z- transform of its impulse response evaluated on the unit circle!  Convolution in time domain is equivalent to multiplication in the frequency domain. Input Complex Coefficient

Sinusoidal Response Example For sinusoidal input: And system described by: Derive an expression for and plot the frequency response (phase and magnitude) of the system output.

Homework(2) For sinusoidal inputs: And system described by: Plot the frequency response (phase and magnitude) and compute the corresponding outputs.

Sampling Sampling rate determines the highest signal frequency that can be reconstructed from the signal samples without error. At least 3 samples (2 complete sampling intervals) must fall within a period for digitization without aliasing. In other words the sampling rate must be greater than twice the highest signal frequency for a band limited signal. F s =200 Hz T s = 5 ms

Sampling Aliasing I – The Movie (FS=200, Range Hz) Run following mfile at:

Sampling Aliasing II – The Sequel (FS=200, Range Hz). Change beginning and ending frequency parameters in following mfile to run:

The Sampling Theorem A band-limited continuous signal s(t) can be reconstructed without error from its samples provided: where f s is the sampling frequency in samples per second, and f b is the frequency above which s(t) has no energy.

Aliased Signal Spectra Sampling in time with sampling frequency f s creates an infinite pattern of a shifted analog spectra so that frequency domain has a periodicity of f s. Let S a (f) be the spectra of the original analog signal, the spectrum of the sampled signal becomes:

Aliased Signal Spectra Spectral periodicity of a low-pass signal (not really band- limited) resulting from an 8 kHz sampling

Restoring Sampled Signals A sampled signal is reconstructed by low-pass filtering the samples with a cut-off near the folding frequency

Aliased Signal Example Before sampling at a given rate, signals are often low-pass filtered (anti-aliasing filter) to limit distortions from aliasing.  Original Sound  Limited Bandwidth (LPF with 900 Hz cutoff) and sampled at 2 kHz  Original Sound sampled at 2 kHz (aliasing)

Homework(3) Determine the aliased frequencies in the range of For the following sampling frequency and signal pairs:

Linear Difference Equations Discrete systems are described by Z-transforms in the frequency domain and difference equations in the time domain. Digital filters can be designed in either domain. Find the impulse response of the following filters. (FIR) (IIR) a) Compute impulse response directly by hand b) Use Matlab function “filter” c) Take inverse of Z transform d) Examine poles and zeros of filters

Linear Difference Equations (FIR)

Linear Difference Equations (IIR)

Homework (4) Find the impulse response of the following filters. a) b) c) 1) Use Matlab function “filter” 2) Take inverse of Z transform 3) Examine poles of filter and comment on expected stability