Sampling and Reconstruction of Analog Signals

Similar documents
Islamic University of Gaza. Faculty of Engineering Electrical Engineering Department Spring-2011

Digital Signal Processing

(i) Understanding of the characteristics of linear-phase finite impulse response (FIR) filters

Sampling of Continuous-Time Signals. Reference chapter 4 in Oppenheim and Schafer.

Lecture Schedule: Week Date Lecture Title

(i) Understanding of the characteristics of linear-phase finite impulse response (FIR) filters

Music 270a: Fundamentals of Digital Audio and Discrete-Time Signals

Lecture 7 Frequency Modulation

Digital Processing of Continuous-Time Signals

Module 3 : Sampling and Reconstruction Problem Set 3

Figure 1: Block diagram of Digital signal processing

Digital Processing of

(i) Understanding the basic concepts of signal modeling, correlation, maximum likelihood estimation, least squares and iterative numerical methods

PYKC 27 Feb 2017 EA2.3 Electronics 2 Lecture PYKC 27 Feb 2017 EA2.3 Electronics 2 Lecture 11-2

Design IV. E232 Spring 07

Advanced Digital Signal Processing Part 2: Digital Processing of Continuous-Time Signals

!"!#"#$% Lecture 2: Media Creation. Some materials taken from Prof. Yao Wang s slides RECAP

Continuous vs. Discrete signals. Sampling. Analog to Digital Conversion. CMPT 368: Lecture 4 Fundamentals of Digital Audio, Discrete-Time Signals

Multirate Digital Signal Processing

CMPT 318: Lecture 4 Fundamentals of Digital Audio, Discrete-Time Signals

Laboratory Assignment 5 Amplitude Modulation

Laboratory Assignment 2 Signal Sampling, Manipulation, and Playback

Experiment 8: Sampling

PROBLEM SET 6. Note: This version is preliminary in that it does not yet have instructions for uploading the MATLAB problems.

Concordia University. Discrete-Time Signal Processing. Lab Manual (ELEC442) Dr. Wei-Ping Zhu

Discrete-Time Signal Processing (DTSP) v14

Signal Characteristics

II Year (04 Semester) EE6403 Discrete Time Systems and Signal Processing

Multirate DSP, part 1: Upsampling and downsampling

Moving from continuous- to discrete-time

IIR Filter Design Chapter Intended Learning Outcomes: (i) Ability to design analog Butterworth filters

Pulse Code Modulation (PCM)

EE 215 Semester Project SPECTRAL ANALYSIS USING FOURIER TRANSFORM

Sampling and Signal Processing

Final Exam Practice Questions for Music 421, with Solutions

The quality of the transmission signal The characteristics of the transmission medium. Some type of transmission medium is required for transmission:

Lecture 17 z-transforms 2

Lecture 3 Review of Signals and Systems: Part 2. EE4900/EE6720 Digital Communications

ANALOGUE AND DIGITAL COMMUNICATION

ece 429/529 digital signal processing robin n. strickland ece dept, university of arizona ECE 429/529 RNS

Digital Signal Processing (Subject Code: 7EC2)

ECE 556 BASICS OF DIGITAL SPEECH PROCESSING. Assıst.Prof.Dr. Selma ÖZAYDIN Spring Term-2017 Lecture 2

Chapter 2: Digitization of Sound

ECE 2111 Signals and Systems Spring 2012, UMD Experiment 9: Sampling

Multirate DSP, part 3: ADC oversampling

Signals and Systems. Lecture 13 Wednesday 6 th December 2017 DR TANIA STATHAKI

Recall. Sampling. Why discrete time? Why discrete time? Many signals are continuous-time signals Light Object wave CCD

ECE 484 Digital Image Processing Lec 09 - Image Resampling

ECEGR Lab #8: Introduction to Simulink

SAMPLING THEORY. Representing continuous signals with discrete numbers

Nyquist's criterion. Spectrum of the original signal Xi(t) is defined by the Fourier transformation as follows :

Brief Introduction to Signals & Systems. Phani Chavali

LABORATORY - FREQUENCY ANALYSIS OF DISCRETE-TIME SIGNALS

Final Exam Solutions June 14, 2006

THE CITADEL THE MILITARY COLLEGE OF SOUTH CAROLINA. Department of Electrical and Computer Engineering. ELEC 423 Digital Signal Processing

ECE438 - Laboratory 7a: Digital Filter Design (Week 1) By Prof. Charles Bouman and Prof. Mireille Boutin Fall 2015

Waveform Encoding - PCM. BY: Dr.AHMED ALKHAYYAT. Chapter Two

Infocommunication. Sampling, Quantization. - Bálint TÓTH, BME TMIT -

Chapter-2 SAMPLING PROCESS

Multirate Signal Processing Lecture 7, Sampling Gerald Schuller, TU Ilmenau

The University of Texas at Austin Dept. of Electrical and Computer Engineering Final Exam

PROBLEM SET 5. Reminder: Quiz 1will be on March 6, during the regular class hour. Details to follow. z = e jω h[n] H(e jω ) H(z) DTFT.

ECE 429 / 529 Digital Signal Processing

Digital Filters IIR (& Their Corresponding Analog Filters) Week Date Lecture Title

Signals and Systems Using MATLAB

Biomedical Signals. Signals and Images in Medicine Dr Nabeel Anwar

Electrical & Computer Engineering Technology

Signal Processing Toolbox

NON-UNIFORM SIGNALING OVER BAND-LIMITED CHANNELS: A Multirate Signal Processing Approach. Omid Jahromi, ID:

Topic 2. Signal Processing Review. (Some slides are adapted from Bryan Pardo s course slides on Machine Perception of Music)

Armstrong Atlantic State University Engineering Studies MATLAB Marina Sound Processing Primer

Lecture 2 Review of Signals and Systems: Part 1. EE4900/EE6720 Digital Communications

The Sampling Theorem:

Discrete-time Signals & Systems

TE 302 DISCRETE SIGNALS AND SYSTEMS. Chapter 1: INTRODUCTION

SIGNALS AND SYSTEMS LABORATORY 13: Digital Communication

Basic Signals and Systems

Overview of Signal Processing

Sistemas de Aquisição de Dados. Mestrado Integrado em Eng. Física Tecnológica 2015/16 Aula 3-29 de Setembro

EE 230 Lecture 39. Data Converters. Time and Amplitude Quantization

Laboratory Assignment 4. Fourier Sound Synthesis

B.Tech III Year II Semester (R13) Regular & Supplementary Examinations May/June 2017 DIGITAL SIGNAL PROCESSING (Common to ECE and EIE)

Digital Video and Audio Processing. Winter term 2002/ 2003 Computer-based exercises

Annex. 1.3 Measuring information

Filter Banks I. Prof. Dr. Gerald Schuller. Fraunhofer IDMT & Ilmenau University of Technology Ilmenau, Germany. Fraunhofer IDMT

Signal Processing. Introduction

Final Exam Solutions June 7, 2004

Outline. Discrete time signals. Impulse sampling z-transform Frequency response Stability INF4420. Jørgen Andreas Michaelsen Spring / 37 2 / 37

Terminology (1) Chapter 3. Terminology (3) Terminology (2) Transmitter Receiver Medium. Data Transmission. Direct link. Point-to-point.

Cyber-Physical Systems ADC / DAC

Chapter 3. Data Transmission

DSP First Lab 08: Frequency Response: Bandpass and Nulling Filters

y(n)= Aa n u(n)+bu(n) b m sin(2πmt)= b 1 sin(2πt)+b 2 sin(4πt)+b 3 sin(6πt)+ m=1 x(t)= x = 2 ( b b b b

DIGITAL SIGNAL PROCESSING. Chapter 1 Introduction to Discrete-Time Signals & Sampling

Overview of Digital Signal Processing

Sampling, interpolation and decimation issues

Signal Sampling. Sampling. Sampling. Sampling. Sampling. Sampling

Chapter 3 Data Transmission COSC 3213 Summer 2003

Filters. Materials from Prof. Klaus Mueller

Analog-Digital Interface

Transcription:

Sampling and Reconstruction of Analog Signals Chapter Intended Learning Outcomes: (i) Ability to convert an analog signal to a discrete-time sequence via sampling (ii) Ability to construct an analog signal from a discrete-time sequence (iii) Understanding the conditions when a sampled signal can uniquely represent its analog counterpart H. C. So Page 1 Semester A, 2017-2018

Sampling Process of converting a continuous-time signal discrete-time sequence into a is obtained by extracting every s where is known as the sampling period or interval sample at analog discrete-time signal signal Fig.4.1: Conversion of analog signal to discrete-time sequence Relationship between and is: (4.1) H. C. So Page 2 Semester A, 2017-2018

Conceptually, conversion of to is achieved by a continuous-time to discrete-time (CD) converter: CD converter impulse train to sequence conversion t n Fig.4.2: Block diagram of CD converter H. C. So Page 3 Semester A, 2017-2018

A fundamental question is whether can uniquely represent or if we can use to reconstruct t Fig.4.3: Different analog signals map to same sequence H. C. So Page 4 Semester A, 2017-2018

But, the answer is yes when: (1) is bandlimited such that its Fourier transform for where is called the bandwidth (2) Sampling period is sufficiently small Example 4.1 The continuous-time signal is used as the input for a CD converter with the sampling period s. Determine the resultant discrete-time signal. According to (4.1), is The frequency in is while that of is H. C. So Page 5 Semester A, 2017-2018

Frequency Domain Representation of Sampled Signal In the time domain, is obtained by multiplying by the impulse train : (4.2) with the use of the sifting property of (2.12) Let the sampling frequency in radian be in Hz). From Example 2.8: (or (4.3) H. C. So Page 6 Semester A, 2017-2018

Using multiplication property of Fourier transform: (4.4) where the convolution operation corresponds to continuoustime signals Using (4.2)-(4.4) and properties of, is: H. C. So Page 7 Semester A, 2017-2018

(4.5) which is the sum of infinite copies of scaled by H. C. So Page 8 Semester A, 2017-2018

When can get is chosen sufficiently large such that all copies of do not overlap, that is, or, we from............ Fig.4.4: for sufficiently large H. C. So Page 9 Semester A, 2017-2018

When is not chosen sufficiently large such that, copies of overlap, we cannot get from, which is referred to aliasing............ Fig.4.5: when is not large enough H. C. So Page 10 Semester A, 2017-2018

Nyquist Sampling Theorem (1928) Let be a bandlimited continuous-time signal with (4.6) Then is uniquely determined by its samples,, if (4.7) The bandwidth is also known as the Nyquist frequency while is called the Nyquist rate and must exceed it in order to avoid aliasing H. C. So Page 11 Semester A, 2017-2018

Application Biomedical Hz 1 khz Telephone speech khz 8 khz Music khz 44.1 khz Ultrasonic khz 250 khz Radar MHz 200 MHz Table 4.1: Typical bandwidths and sampling frequencies in signal processing applications Example 4.2 Determine the Nyquist frequency and Nyquist rate for the continuous-time signal which has the form of: The frequencies are 0, and. The Nyquist frequency is and the Nyquist rate is H. C. So Page 12 Semester A, 2017-2018

...... Fig.4.6: Multiplying and to recover In frequency domain, we multiply by with amplitude and bandwidth with, to obtain, and it corresponds to H. C. So Page 13 Semester A, 2017-2018

Reconstruction Process of transforming back to DC converter sequence to impulse train conversion Fig.4.7: Block diagram of DC converter From Fig.4.6, is (4.8) where, which is a lowpass filter H. C. So Page 14 Semester A, 2017-2018

For simplicity, we set as the average of and : (4.9) To get, we take inverse Fourier transform of and use Example 2.5: (4.10) where H. C. So Page 15 Semester A, 2017-2018

Using (2.23)-(2.24), (4.2) and (2.11)-(2.12), is: (4.11) which holds for all real values of H. C. So Page 16 Semester A, 2017-2018

The interpolation formula can be verified at : (4.12) It is easy to see that (4.13) For, we use s rule to obtain: (4.14) Substituting (4.13)-(4.14) into (4.12) yields: (4.15) which aligns with H. C. So Page 17 Semester A, 2017-2018

Example 4.3 Given a discrete-time sequence. Generate its time-delayed version which has the form of where and is a positive integer. Applying (4.11) with : By employing a change of variable of : Is it practical to get y[n]? H. C. So Page 18 Semester A, 2017-2018

Note that when, the time-shifted signal is simply obtained by shifting the sequence by samples: Sampling and Reconstruction in Digital Signal Processing CD converter digital signal processor DC converter Fig.4.8: Ideal digital processing of analog signal CD converter produces a sequence from is processed in discrete-time domain to give DC converter creates from according to (4.11): (4.16) H. C. So Page 19 Semester A, 2017-2018

anti-aliasing filter analog-to-digital converter digital signal processor digital-to-analog converter Fig.4.9: Practical digital processing of analog signal may not be precisely bandlimited a lowpass filter or anti-aliasing filter is needed to process Ideal CD converter is approximated by AD converter Not practical to generate AD converter introduces quantization error Ideal DC converter is approximated by DA converter because ideal reconstruction of (4.16) is impossible Not practical to perform infinite summation Not practical to have future data and are quantized signals H. C. So Page 20 Semester A, 2017-2018

Example 4.4 Suppose a continuous-time signal is sampled at a sampling frequency of 1000Hz to produce : Determine 2 possible positive values of, say, and. Discuss if or will be obtained when passing through the DC converter. According to (4.1) with s: is easily computed as: H. C. So Page 21 Semester A, 2017-2018

can be obtained by noting the periodicity of a sinusoid: As a result, we have: This is illustrated using the MATLAB code: O1=250*pi; %first frequency O2=2250*pi; %second frequency Ts=1/100000;%successive sample separation is 0.01T t=0:ts:0.02;%observation interval x1=cos(o1.*t); %tone from first frequency x2=cos(o2.*t); %tone from second frequency There are 2001 samples in 0.02s and interpolating the successive points based on plot yields good approximations H. C. So Page 22 Semester A, 2017-2018

1 0.8 0.6 0.4 0.2 0-0.2-0.4 x[n] -0.6-0.8-1 0 5 10 15 20 n Fig.4.10: Discrete-time sinusoid H. C. So Page 23 Semester A, 2017-2018

1 0.8 0.6 0.4 0.2 0-0.2-0.4-0.6-0.8 Ω 1 Ω 2-1 0 0.005 0.01 0.015 0.02 t Fig.4.11: Continuous-time sinusoids H. C. So Page 24 Semester A, 2017-2018

Passing but not through the DC converter only produces The Nyquist frequency of is and hence the sampling frequency without aliasing is Given Hz or, does not correspond to We can recover because the Nyquist frequency and Nyquist rate for are and Based on (4.11), is: with s H. C. So Page 25 Semester A, 2017-2018

The MATLAB code for reconstructing n=-10:30; %add 20 past and future samples x=cos(pi.*n./4); T=1/1000; %sampling interval is 1/1000 for l=1:2000 %observed interval is [0,0.02] t=(l-1)*t/100;%successive sample separation is 0.01T h=sinc((t-n.*t)./t); xr(l)=x*h.'; %approximate interpolation of (4.11) end is: We compute 2000 samples of in s The value of each at time t is approximated as x*h.' where the sinc vector is updated for each computation The MATLAB program is provided as ex4_4.m H. C. So Page 26 Semester A, 2017-2018

1 0.8 0.6 0.4 0.2 0-0.2-0.4-0.6-0.8 x r (t) -1 0 0.005 0.01 0.015 0.02 t Fig.4.12: Reconstructed continuous-time sinusoid H. C. So Page 27 Semester A, 2017-2018

Example 4.5 Play the sound for a discrete-time tone using MATLAB. The frequency of the corresponding analog signal is 440 Hz which corresponds to the A note in the American Standard pitch. The sampling frequency is 8000 Hz and the signal has a duration of 0.5 s. The MATLAB code is A=sin(2*pi*440*(0:1/8000:0.5));%discrete-time A sound(a,8000); %DA conversion and play Note that sampling frequency in Hz is assumed for sound. The frequencies of notes B, C#, D, E and F# are 493.88 Hz, 554.37 Hz, 587.33 Hz, 659.26 Hz and 739.99 Hz, respectively. You can easily produce a piece of music with notes: A, A, E, E, F#, F#, E, E, D, D, C#, C#, B, B, A, A. H. C. So Page 28 Semester A, 2017-2018