Basics on Digital Signal Processing

Total Page:16

File Type:pdf, Size:1020Kb

Basics on Digital Signal Processing Basics on Digital Signal Processing z - transform - Digital Filters Vassilis Anastassopoulos Electronics Laboratory, Physics Department, University of Patras Outline of the Lecture 1. The z-transform 2. Properties 3. Examples 4. Digital filters 5. Design of IIR and FIR filters 6. Linear phase 2/31 z-Transform 2 N 1 N 1 j nk n X (k) x(n) e N X (z) x(n) z n0 n0 Time to frequency Time to z-domain (complex) Transformation tool is the Transformation tool is complex wave z e j e j j j2k / N e e With amplitude ρ changing(?) with time With amplitude |ejω|=1 The z-transform is more general than the DFT 3/31 z-Transform For z=ejω i.e. ρ=1 we work on the N 1 unit circle X (z) x(n) z n And the z-transform degenerates n0 into the Fourier transform. I -z z=R+jI |z|=1 R The DFT is an expression of the z-transform on the unit circle. The quantity X(z) must exist with finite value on the unit circle i.e. must posses spectrum with which we can describe a signal or a system. 4/31 z-Transform convergence We are interested in those values of z for which X(z) converges. This region should contain the unit circle. I Why is it so? |a| N 1 N 1 X (z) x(n) zn x(n) (1/zn ) n0 n0 R At z=0, X(z) diverges ROC The values of z for which X(z) diverges are called poles of X(z). 5/31 z-Transform example Which is the z-transform and the ROC of a discrete time sequence x(n)=an for n0 and a<1 ? x(n) 1 a I a2 a3 |a|a 4 n R N 1 X (z) x(n) zn ROCan z n (az1)n n0 n0 n0 -1 1 z which for |az |<1 or |z|>|a| converges to X (z) 1 az1 z a The pole z=a, is never included in the ROC 6/31 Poles and zeros of X(z) Poles: X(z)= Zeros: X(z)=0 For infinite sequences, X(z) converges everywhere outside the circle with radius the poleI with maximum value. For finite sequences, X(z) converges everywhere |a| except at z=0 x(n) 5 R 3 3 X(z) x(n)z n z 1 3z 2 5z 3 3z 4 z 5 1 1 n n ROC For stable, causal digital systems the region of convergence includes the Unit Circle so that the system possesses spectrum 7/31 z-Transform general form a a z1 a z2 a zN k(z z )(z z )(z z ) 0 1 2 N X (z) 1 2 N X (z) 1 2 M b0 b1z b2z bN z (z p1)(z p2 )(z pN ) Those values of z (zi) that make the nominator zero are called zeros. While the poles are the values of z (pi) that make the denominator zero and thus X(z) diverges. For stable, causal digital systems the region of convergence includes the Unit Circle so that the system possesses spectrum H(e jT ) H(z) h(n)z n h(n)e jT ze jT n ze jT n 8/31 Inverse z-Transform A simple way to evaluate the signal from the X(z) is to perform the division 1 2z 1 z 2 X(z) 1 z 1 0.356z 2 X(z) 1 3z 1 3.6439z 2 2.5756z 3 The signal is x(0)=1, x(1)=3, x(2)=3.6439, x(3)=2.5756 9/31 z-Transform properties x1(n) X1(z) Linearity x2 (n) X 2 (z) ax1(n) bx2 (n) aX1(z) bX 2 (z) x(n) X (z) Delay or Shift x(n m) zm X (z) Convolution y(n) h(k)x(n k) Y(z) H(z)X(z) k 10/31 z-Transform and digital systems N M y(n) ak x(n k) bk y(n k) k0 k1 x(n) x(n-1) x(n-2) x(n-Ν+1) x(n-Ν) Ts Ts Ts a0 a1 a2 aΝ-1 aΝ y(n) N + k ak z k0 H (z) N bΜ bΜ-1 b2 b1 k 1 bk z k1 T T T y(n-Μ) s y(n-Μ+1) y(n-2) s y(n-1) s 11/31 Digital system z-Transform derivation N M y(n) ak x(n k) bk y(n k) z-transform both sides k0 k1 N M n n n Interchange z y(n) z ak x(n k) z bk y(n k) summations n0 n0 k0 n0 k1 N M n n Y (z) ak z x(n k) bk z y(n k) Replace transforms k0 n0 k1 n0 N M Y (z) ak X (z k) bkY (z k) Shift property k0 k1 N M k k Common factors Y(z) ak z X(z) bk z Y(z) k0 k1 N M k k Y (z) X (z)ak z Y (z)bk z k0 k1 12/31 Digital system z-Transform derivation M N N M k y(nk) aCommonx(n k factors) b y(n k) Y (z) Y (z)bk z X (z)ak z k k k1 k0 k0 k1 M N x(n) x(n-1) x(n-2) x(n-Ν+1) x(n-Ν) Ts Ts k Ts k Y (z)(1 bk z ) X (z)ak z k1 k0 a0 a1 a2 aΝ-1 aΝ N k ak z Y (z) y(n) k0 Frequency N + H(z) X (z) k 1 bk z k1 bΜ bΜ-1 b2 b1 Ts Ts Ts Time y(n-Μ) y(n-Μ+1) y(n-2) y(n-1) 13/31 Application Find the impulse response h(n) and the input-output relationship of the filter described by 1 z 1 H(z) 1 0.5z 1 Examine the stability of the filter and find its frequency response. Solution -1 Y(z) 1 z 1 x(n) z H(z) X(z) 1 0.5z 1 1 1 Z1 -1 Y(z) 0.5Y(z)z X(z) X(z)z y(n) y(n) 0.5y(n 1) x(n) x(n 1) -0.5 y(n) x(n) x(n 1) 0.5y(n 1) z-1 h(n) is obtained from the terms of the polynomial which results after the division (1 z 1 ) / (1 0.5z 1 ) 11.5z 1 0.75z 2 0.375z 3 14/31 Application 1 Im 1 z H(z) Stable 1 0.5z 1 |z|=1 1 Re Frequency response using MATLAB 5 2 4.5 1.8 4 1.6 3.5 1.4 3 1.2 2.5 1 Phase Magnitude 2 0.8 1.5 0.6 1 0.4 0.5 0.2 0 0 0 0.1 0.2 0.3 0.4 0 0.1 0.2 0.3 0.4 Frequency Frequency 15/31 Frequency on the Unit Circle π/2 or Fs/4 Im 1 z 1 H(z) z=e1jω0t .5z 1 |z|=1 π or Fs/2 0 Re The region from 0 to π corresponds to the region 0- Fs/2 of the Frequency response 16/31 Digital Filters •They are characterized by their Impulse Response h(n), their Transfer Function H(z) and their Frequency Response H(ω). •They can have memory, high accuracy and no drift with time and temperature. •They can possess linear phase. •They can be implemented by digital computers. 17/31 Digital Filters - Categories N M IIR y(n) a(k)x(n k) bk y(n k) k0 k1 N k ak z x(n) x(n-1) x(n-2) x(n-Ν+1) x(n-Ν) Ts Ts Ts k0 H(z) M 1 b z k k a a a a a k1 0 1 2 N-1 N y(n) + bΜ bΜ-1 b2 b1 T T T y(n-Μ) s y(n-Μ+1) y(n-2) s y(n-1) s 18/31 Digital Filters - Categories N N y(n) h(k)x(n k) a(k)x(n k) FIR k0 k0 N N k k H(z) ak z h(k)z k0 k0 x(n) x(n-1) x(n-2) x(n-Ν+2) x(n-Ν+1) Ts Ts Ts aΝ-2 ή a ή a0 ή a1 ή a2 ή Ν-1 h(0) h(1) h(2) h(Ν-2) h(Ν-1) y(n) + •Stable •Linear phase 19/31 Digital Filters - Examples a a z 1 a z 2 0 1 2 1.5 1 H(z) 1 2 1 b1z b2 z 0 1 a0=0.498, a1=0.927, -1 a2=0.498, b1=-0.674 b2=-0.363. 0.5 IIR Phase Magnitude IIR Magnitude -2 0 -3 0 0.1 0.2 0.3 0.4 0 0.1 0.2 0.3 0.4 Frequency Frequency 11 1.5 4 H(z) h(k)z k 2 k0 1 0 h(1)=h(10)=-0.04506 0.5 FIR Phase h(2)=h(9)=0.06916 FIR Magnitude -2 h(3)=h(8)=-0.0553 0 -4 0 0.1 0.2 0.3 0.4 0 0.1 0.2 0.3 0.4 h(4)=h(7)=-.06342 Frequency Frequency h(5)=h(4)=0.5789 20/31 IIR filter design Design using the Bilinear z-transform, BZT Design using the position of poles and zeros on the unit circle Fs/4 Im •The frequency response is zero at the points of zeros |z|=1 Fs/2 1 0 or Fs •The frequency response takes a peak at the position of poles.
Recommended publications
  • Detection and Estimation Theory Introduction to ECE 531 Mojtaba Soltanalian- UIC the Course
    Detection and Estimation Theory Introduction to ECE 531 Mojtaba Soltanalian- UIC The course Lectures are given Tuesdays and Thursdays, 2:00-3:15pm Office hours: Thursdays 3:45-5:00pm, SEO 1031 Instructor: Prof. Mojtaba Soltanalian office: SEO 1031 email: [email protected] web: http://msol.people.uic.edu/ The course Course webpage: http://msol.people.uic.edu/ECE531 Textbook(s): * Fundamentals of Statistical Signal Processing, Volume 1: Estimation Theory, by Steven M. Kay, Prentice Hall, 1993, and (possibly) * Fundamentals of Statistical Signal Processing, Volume 2: Detection Theory, by Steven M. Kay, Prentice Hall 1998, available in hard copy form at the UIC Bookstore. The course Style: /Graduate Course with Active Participation/ Introduction Let’s start with a radar example! Introduction> Radar Example QUIZ Introduction> Radar Example You can actually explain it in ten seconds! Introduction> Radar Example Applications in Transportation, Defense, Medical Imaging, Life Sciences, Weather Prediction, Tracking & Localization Introduction> Radar Example The strongest signals leaking off our planet are radar transmissions, not television or radio. The most powerful radars, such as the one mounted on the Arecibo telescope (used to study the ionosphere and map asteroids) could be detected with a similarly sized antenna at a distance of nearly 1,000 light-years. - Seth Shostak, SETI Introduction> Estimation Traditionally discussed in STATISTICS. Estimation in Signal Processing: Digital Computers ADC/DAC (Sampling) Signal/Information Processing Introduction> Estimation The primary focus is on obtaining optimal estimation algorithms that may be implemented on a digital computer. We will work on digital signals/datasets which are typically samples of a continuous-time waveform.
    [Show full text]
  • Z-Transformation - I
    Digital signal processing: Lecture 5 z-transformation - I Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/1 Review of last lecture • Fourier transform & inverse Fourier transform: – Time domain & Frequency domain representations • Understand the “true face” (latent factor) of some physical phenomenon. – Key points: • Definition of Fourier transformation. • Definition of inverse Fourier transformation. • Condition for a signal to have FT. Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/2 Review of last lecture • The sampling theorem – If the highest frequency component contained in an analog signal is f0, the sampling frequency (the Nyquist rate) should be 2xf0 at least. • Frequency response of an LTI system: – The Fourier transformation of the impulse response – Physical meaning: frequency components to keep (~1) and those to remove (~0). – Theoretic foundation: Convolution theorem. Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/3 Topics of this lecture Chapter 6 of the textbook • The z-transformation. • z-変換 • Convergence region of z- • z-変換の収束領域 transform. -変換とフーリエ変換 • Relation between z- • z transform and Fourier • z-変換と差分方程式 transform. • 伝達関数 or システム • Relation between z- 関数 transform and difference equation. • Transfer function (system function). Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/4 z-transformation(z-変換) • Laplace transformation is important for analyzing analog signal/systems. • z-transformation is important for analyzing discrete signal/systems. • For a given signal x(n), its z-transformation is defined by ∞ Z[x(n)] = X (z) = ∑ x(n)z−n (6.3) n=0 where z is a complex variable. Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/5 One-sided z-transform and two-sided z-transform • The z-transform defined above is called one- sided z-transform (片側z-変換).
    [Show full text]
  • 3F3 – Digital Signal Processing (DSP)
    3F3 – Digital Signal Processing (DSP) Simon Godsill www-sigproc.eng.cam.ac.uk/~sjg/teaching/3F3 Course Overview • 11 Lectures • Topics: – Digital Signal Processing – DFT, FFT – Digital Filters – Filter Design – Filter Implementation – Random signals – Optimal Filtering – Signal Modelling • Books: – J.G. Proakis and D.G. Manolakis, Digital Signal Processing 3rd edition, Prentice-Hall. – Statistical digital signal processing and modeling -Monson H. Hayes –Wiley • Some material adapted from courses by Dr. Malcolm Macleod, Prof. Peter Rayner and Dr. Arnaud Doucet Digital Signal Processing - Introduction • Digital signal processing (DSP) is the generic term for techniques such as filtering or spectrum analysis applied to digitally sampled signals. • Recall from 1B Signal and Data Analysis that the procedure is as shown below: • is the sampling period • is the sampling frequency • Recall also that low-pass anti-aliasing filters must be applied before A/D and D/A conversion in order to remove distortion from frequency components higher than Hz (see later for revision of this). • Digital signals are signals which are sampled in time (“discrete time”) and quantised. • Mathematical analysis of inherently digital signals (e.g. sunspot data, tide data) was developed by Gauss (1800), Schuster (1896) and many others since. • Electronic digital signal processing (DSP) was first extensively applied in geophysics (for oil-exploration) then military applications, and is now fundamental to communications, mobile devices, broadcasting, and most applications of signal and image processing. There are many advantages in carrying out digital rather than analogue processing; among these are flexibility and repeatability. The flexibility stems from the fact that system parameters are simply numbers stored in the processor.
    [Show full text]
  • CONVOLUTION: Digital Signal Processing .R .Hamming, W
    CONVOLUTION: Digital Signal Processing AN-237 National Semiconductor CONVOLUTION: Digital Application Note 237 Signal Processing January 1980 Introduction As digital signal processing continues to emerge as a major Decreasing the pulse width while increasing the pulse discipline in the field of electrical engineering, an even height to allow the area under the pulse to remain constant, greater demand has evolved to understand the basic theo- Figure 1c, shows from eq(1) and eq(2) the bandwidth or retical concepts involved in the development of varied and spectral-frequency content of the pulse to have increased, diverse signal processing systems. The most fundamental Figure 1d. concepts employed are (not necessarily listed in the order Further altering the pulse to that of Figure 1e provides for an [ ] of importance) the sampling theorem 1 , Fourier transforms even broader bandwidth, Figure 1f. If the pulse is finally al- [ ][] 2 3, convolution, covariance, etc. tered to the limit, i.e., the pulsewidth being infinitely narrow The intent of this article will be to address the concept of and its amplitude adjusted to still maintain an area of unity convolution and to present it in an introductory manner under the pulse, it is found in 1g and 1h the unit impulse hopefully easily understood by those entering the field of produces a constant, or ``flat'' spectrum equal to 1 at all digital signal processing. frequencies. Note that if ATe1 (unit area), we get, by defini- It may be appropriate to note that this article is Part II (Part I tion, the unit impulse function in time.
    [Show full text]
  • The Scientist and Engineer's Guide to Digital Signal Processing Properties of Convolution
    CHAPTER 7 Properties of Convolution A linear system's characteristics are completely specified by the system's impulse response, as governed by the mathematics of convolution. This is the basis of many signal processing techniques. For example: Digital filters are created by designing an appropriate impulse response. Enemy aircraft are detected with radar by analyzing a measured impulse response. Echo suppression in long distance telephone calls is accomplished by creating an impulse response that counteracts the impulse response of the reverberation. The list goes on and on. This chapter expands on the properties and usage of convolution in several areas. First, several common impulse responses are discussed. Second, methods are presented for dealing with cascade and parallel combinations of linear systems. Third, the technique of correlation is introduced. Fourth, a nasty problem with convolution is examined, the computation time can be unacceptably long using conventional algorithms and computers. Common Impulse Responses Delta Function The simplest impulse response is nothing more that a delta function, as shown in Fig. 7-1a. That is, an impulse on the input produces an identical impulse on the output. This means that all signals are passed through the system without change. Convolving any signal with a delta function results in exactly the same signal. Mathematically, this is written: EQUATION 7-1 The delta function is the identity for ( ' convolution. Any signal convolved with x[n] *[n] x[n] a delta function is left unchanged. This property makes the delta function the identity for convolution. This is analogous to zero being the identity for addition (a%0 ' a), and one being the identity for multiplication (a×1 ' a).
    [Show full text]
  • Exploring Anti-Aliasing Filters in Signal Conditioners for Mixed-Signal, Multimodal Sensor Conditioning by Arun T
    Texas Instruments Incorporated Amplifiers: Op Amps Exploring anti-aliasing filters in signal conditioners for mixed-signal, multimodal sensor conditioning By Arun T. Vemuri Systems Architect, Enhanced Industrial Introduction Figure 1. Multimodal, mixed-signal sensor-signal conditioner Some sensor-signal conditioners are used to process the output Analog Digital of multiple sense elements. This Domain Domain processing is often provided by multimodal, mixed-signal condi- tioners that can handle the out- puts from several sense elements Sense Digital Amplifier 1 ADC 1 at the same time. This article Element 1 Filter 1 analyzes the operation of anti- Processed aliasing filters in such sensor- Intelligent Output Compensation signal conditioners. Sense Digital Basics of sensor-signal Amplifier 2 ADC 2 Element 2 Filter 2 conditioners Sense elements, or transducers, convert a physical quantity of interest into electrical signals. Examples include piezo resistive bridges used to measure pres- sure, piezoelectric transducers used to detect ultrasonic than one sense element is processed by the same signal waves, and electrochemical cells used to measure gas conditioner is called multimodal signal conditioning. concentrations. The electrical signals produced by sense Mixed-signal signal conditioning elements are small and exhibit nonidealities, such as tem- Another aspect of sensor-signal conditioning is the electri- perature drifts and nonlinear transfer functions. cal domain in which the signal conditioning occurs. TI’s Sensor analog front ends such as the Texas Instruments PGA309 is an example of a device where the signal condi- (TI) LMP91000 and sensor-signal conditioners such as TI’s tioning of resistive-bridge sense elements occurs in the PGA400/450 are used to amplify the small signals produced analog domain.
    [Show full text]
  • Digital Signal Processing: Theory and Practice, Hardware and Software
    AC 2009-959: DIGITAL SIGNAL PROCESSING: THEORY AND PRACTICE, HARDWARE AND SOFTWARE Wei PAN, Idaho State University Wei Pan is Assistant Professor and Director of VLSI Laboratory, Electrical Engineering Department, Idaho State University. She has several years of industrial experience including Siemens (project engineering/management.) Dr. Pan is an active member of ASEE and IEEE and serves on the membership committee of the IEEE Education Society. S. Hossein Mousavinezhad, Idaho State University S. Hossein Mousavinezhad is Professor and Chair, Electrical Engineering Department, Idaho State University. Dr. Mousavinezhad is active in ASEE and IEEE and is an ABET program evaluator. Hossein is the founding general chair of the IEEE International Conferences on Electro Information Technology. Kenyon Hart, Idaho State University Kenyon Hart is Specialist Engineer and Associate Lecturer, Electrical Engineering Department, Idaho State University, Pocatello, Idaho. Page 14.491.1 Page © American Society for Engineering Education, 2009 Digital Signal Processing, Theory/Practice, HW/SW Abstract Digital Signal Processing (DSP) is a course offered by many Electrical and Computer Engineering (ECE) programs. In our school we offer a senior-level, first-year graduate course with both lecture and laboratory sections. Our experience has shown that some students consider the subject matter to be too theoretical, relying heavily on mathematical concepts and abstraction. There are several visible applications of DSP including: cellular communication systems, digital image processing and biomedical signal processing. Authors have incorporated many examples utilizing software packages including MATLAB/MATHCAD in the course and also used classroom demonstrations to help students visualize some difficult (but important) concepts such as digital filters and their design, various signal transformations, convolution, difference equations modeling, signals/systems classifications and power spectral estimation as well as optimal filters.
    [Show full text]
  • ELEG 5173L Digital Signal Processing Ch. 3 Discrete-Time Fourier Transform
    Department of Electrical Engineering University of Arkansas ELEG 5173L Digital Signal Processing Ch. 3 Discrete-Time Fourier Transform Dr. Jingxian Wu [email protected] 2 OUTLINE • The Discrete-Time Fourier Transform (DTFT) • Properties • DTFT of Sampled Signals • Upsampling and downsampling 3 DTFT • Discrete-time Fourier Transform (DTFT) X () x(n)e jn n – (radians): digital frequency • Review: Z-transform: X (z) x(n)zn n0 j X () X (z) | j – Replace z with e . ze • Review: Fourier transform: X () x(t)e jt – (rads/sec): analog frequency 4 DTFT • Relationship between DTFT and Fourier Transform – Sample a continuous time signal x a ( t ) with a sampling period T xs (t) xa (t) (t nT ) xa (nT ) (t nT ) n n – The Fourier Transform of ys (t) jt jnT X s () xs (t)e dt xa (nT)e n – Define: T • : digital frequency (unit: radians) • : analog frequency (unit: radians/sec) – Let x(n) xa (nT) X () X s T 5 DTFT • Relationship between DTFT and Fourier Transform (Cont’d) – The DTFT can be considered as the scaled version of the Fourier transform of the sampled continuous-time signal jt jnT X s () xs (t)e dt xa (nT)e n x(n) x (nT) T a jn X () X s x(n)e T n 6 DTFT • Discrete Frequency – Unit: radians (the unit of continuous frequency is radians/sec) – X ( ) is a periodic function with period 2 j2 n jn j2n jn X ( 2 ) x(n)e x(n)e e x(n)e X () n n n – We only need to consider for • For Fourier transform, we need to consider 1 – f T 2 2T 1 – f T 2 2T 7 DTFT • Example: find the DTFT of the following signal – 1.
    [Show full text]
  • Lecture 12: Sampling, Aliasing, and the Discrete Fourier Transform Foundations of Digital Signal Processing
    Lecture 12: Sampling, Aliasing, and the Discrete Fourier Transform Foundations of Digital Signal Processing Outline • Review of Sampling • The Nyquist-Shannon Sampling Theorem • Continuous-time Reconstruction / Interpolation • Aliasing and anti-Aliasing • Deriving Transforms from the Fourier Transform • Discrete-time Fourier Transform, Fourier Series, Discrete-time Fourier Series • The Discrete Fourier Transform Foundations of Digital Signal Processing Lecture 12: Sampling, Aliasing, and the Discrete Fourier Transform 1 News Homework #5 . Due this week . Submit via canvas Coding Problem #4 . Due this week . Submit via canvas Foundations of Digital Signal Processing Lecture 12: Sampling, Aliasing, and the Discrete Fourier Transform 2 Exam 1 Grades The class did exceedingly well . Mean: 89.3 . Median: 91.5 Foundations of Digital Signal Processing Lecture 12: Sampling, Aliasing, and the Discrete Fourier Transform 3 Lecture 12: Sampling, Aliasing, and the Discrete Fourier Transform Foundations of Digital Signal Processing Outline • Review of Sampling • The Nyquist-Shannon Sampling Theorem • Continuous-time Reconstruction / Interpolation • Aliasing and anti-Aliasing • Deriving Transforms from the Fourier Transform • Discrete-time Fourier Transform, Fourier Series, Discrete-time Fourier Series • The Discrete Fourier Transform Foundations of Digital Signal Processing Lecture 12: Sampling, Aliasing, and the Discrete Fourier Transform 4 Sampling Discrete-Time Fourier Transform Foundations of Digital Signal Processing Lecture 12: Sampling, Aliasing,
    [Show full text]
  • MULTIRATE SIGNAL PROCESSING Multirate Signal Processing
    MULTIRATE SIGNAL PROCESSING Multirate Signal Processing • Definition: Signal processing which uses more than one sampling rate to perform operations • Upsampling increases the sampling rate • Downsampling reduces the sampling rate • Reference: Digital Signal Processing, DeFatta, Lucas, and Hodgkiss B. Baas, EEC 281 431 Multirate Signal Processing • Advantages of lower sample rates –May require less processing –Likely to reduce power dissipation, P = CV2 f, where f is frequently directly proportional to the sample rate –Likely to require less storage • Advantages of higher sample rates –May simplify computation –May simplify surrounding analog and RF circuitry • Remember that information up to a frequency f requires a sampling rate of at least 2f. This is the Nyquist sampling rate. –Or we can equivalently say the Nyquist sampling rate is ½ the sampling frequency, fs B. Baas, EEC 281 432 Upsampling Upsampling or Interpolation •For an upsampling by a factor of I, add I‐1 zeros between samples in the original sequence •An upsampling by a factor I is commonly written I For example, upsampling by two: 2 • Obviously the number of samples will approximately double after 2 •Note that if the sampling frequency doubles after an upsampling by two, that t the original sample sequence will occur at the same points in time t B. Baas, EEC 281 434 Original Signal Spectrum •Example signal with most energy near DC •Notice 5 spectral “bumps” between large signal “bumps” B. Baas, EEC 281 π 2π435 Upsampled Signal (Time) •One zero is inserted between the original samples for 2x upsampling B. Baas, EEC 281 436 Upsampled Signal Spectrum (Frequency) • Spectrum of 2x upsampled signal •Notice the location of the (now somewhat compressed) five “bumps” on each side of π B.
    [Show full text]
  • ESE 531: Digital Signal Processing Lecture Outline ADC Anti-Aliasing
    Lecture Outline ESE 531: Digital Signal Processing ! Data Converters " Anti-aliasing " ADC Lec 12: February 21st, 2017 " Quantization Data Converters, Noise Shaping (con’t) " Practical DAC ! Noise Shaping Penn ESE 531 Spring 2017 - Khanna Penn ESE 531 Spring 2017 - Khanna 2 Anti-Aliasing Filter with ADC ADC Penn ESE 531 Spring 2017 - Khanna 3 Penn ESE 531 Spring 2017 - Khanna 4 Oversampled ADC Oversampled ADC Penn ESE 531 Spring 2017 - Khanna 5 Penn ESE 531 Spring 2017 - Khanna 6 1 Oversampled ADC Oversampled ADC Penn ESE 531 Spring 2017 - Khanna 7 Penn ESE 531 Spring 2017 - Khanna 8 Sampling and Quantization Sampling and Quantization Penn ESE 531 Spring 2017 - Khanna 9 Penn ESE 531 Spring 2017 - Khanna 10 Effect of Quantization Error on Signal Quantization Error ! Quantization error is a deterministic function of the signal " Consequently, the effect of quantization strongly depends on the ! signal itself Model quantization error as noise ! Unless, we consider fairly trivial signals, a deterministic analysis is usually impractical " More common to look at errors from a statistical perspective " "Quantization noise” ! In that case: ! Two aspects " How much noise power (variance) does quantization add to our samples? " How is this noise distributed in frequency? Penn ESE 531 Spring 2017 - Khanna 11 Penn ESE 531 Spring 2017 - Khanna 12 2 Ideal Quantizer Ideal B-bit Quantizer ! Practical quantizers have a limited input range and a finite set of output codes ! Δ Quantization step ! E.g. a 3-bit quantizer can map onto 3 ! Quantization error
    [Show full text]
  • TMS320VC5507/5509 DSP Analog-To-Digital Converter (ADC) Reference Guide
    TMS320VC5507/5509 DSP Analog-to-Digital Converter (ADC) Reference Guide Literature Number: SPRU586B June 2004 IMPORTANT NOTICE Texas Instruments Incorporated and its subsidiaries (TI) reserve the right to make corrections, modifications, enhancements, improvements, and other changes to its products and services at any time and to discontinue any product or service without notice. Customers should obtain the latest relevant information before placing orders and should verify that such information is current and complete. All products are sold subject to TI’s terms and conditions of sale supplied at the time of order acknowledgment. TI warrants performance of its hardware products to the specifications applicable at the time of sale in accordance with TI’s standard warranty. Testing and other quality control techniques are used to the extent TI deems necessary to support this warranty. Except where mandated by government requirements, testing of all parameters of each product is not necessarily performed. TI assumes no liability for applications assistance or customer product design. Customers are responsible for their products and applications using TI components. To minimize the risks associated with customer products and applications, customers should provide adequate design and operating safeguards. TI does not warrant or represent that any license, either express or implied, is granted under any TI patent right, copyright, mask work right, or other TI intellectual property right relating to any combination, machine, or process in which TI products or services are used. Information published by TI regarding third-party products or services does not constitute a license from TI to use such products or services or a warranty or endorsement thereof.
    [Show full text]