Windowing Techniques, the Welch Method for Improvement of Power Spectrum Estimation
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Moving Average Filters
CHAPTER 15 Moving Average Filters The moving average is the most common filter in DSP, mainly because it is the easiest digital filter to understand and use. In spite of its simplicity, the moving average filter is optimal for a common task: reducing random noise while retaining a sharp step response. This makes it the premier filter for time domain encoded signals. However, the moving average is the worst filter for frequency domain encoded signals, with little ability to separate one band of frequencies from another. Relatives of the moving average filter include the Gaussian, Blackman, and multiple- pass moving average. These have slightly better performance in the frequency domain, at the expense of increased computation time. Implementation by Convolution As the name implies, the moving average filter operates by averaging a number of points from the input signal to produce each point in the output signal. In equation form, this is written: EQUATION 15-1 Equation of the moving average filter. In M &1 this equation, x[ ] is the input signal, y[ ] is ' 1 % y[i] j x [i j ] the output signal, and M is the number of M j'0 points used in the moving average. This equation only uses points on one side of the output sample being calculated. Where x[ ] is the input signal, y[ ] is the output signal, and M is the number of points in the average. For example, in a 5 point moving average filter, point 80 in the output signal is given by: x [80] % x [81] % x [82] % x [83] % x [84] y [80] ' 5 277 278 The Scientist and Engineer's Guide to Digital Signal Processing As an alternative, the group of points from the input signal can be chosen symmetrically around the output point: x[78] % x[79] % x[80] % x[81] % x[82] y[80] ' 5 This corresponds to changing the summation in Eq. -
Discrete-Time Fourier Transform 7
Table of Chapters 1. Overview of Digital Signal Processing 2. Review of Analog Signal Analysis 3. Discrete-Time Signals and Systems 4. Sampling and Reconstruction of Analog Signals 5. z Transform 6. Discrete-Time Fourier Transform 7. Discrete Fourier Series and Discrete Fourier Transform 8. Responses of Digital Filters 9. Realization of Digital Filters 10. Finite Impulse Response Filter Design 11. Infinite Impulse Response Filter Design 12. Tools and Applications Chapter 1: Overview of Digital Signal Processing Chapter Intended Learning Outcomes: (i) Understand basic terminology in digital signal processing (ii) Differentiate digital signal processing and analog signal processing (iii) Describe basic digital signal processing application areas H. C. So Page 1 Signal: . Anything that conveys information, e.g., . Speech . Electrocardiogram (ECG) . Radar pulse . DNA sequence . Stock price . Code division multiple access (CDMA) signal . Image . Video H. C. So Page 2 0.8 0.6 0.4 0.2 0 vowel of "a" -0.2 -0.4 -0.6 0 0.005 0.01 0.015 0.02 time (s) Fig.1.1: Speech H. C. So Page 3 250 200 150 100 ECG 50 0 -50 0 0.5 1 1.5 2 2.5 time (s) Fig.1.2: ECG H. C. So Page 4 1 0.5 0 -0.5 transmitted pulse -1 0 0.2 0.4 0.6 0.8 1 time 1 0.5 0 -0.5 received pulse received t -1 0 0.2 0.4 0.6 0.8 1 time Fig.1.3: Transmitted & received radar waveforms H. C. So Page 5 Radar transceiver sends a 1-D sinusoidal pulse at time 0 It then receives echo reflected by an object at a range of Reflected signal is noisy and has a time delay of which corresponds to round trip propagation time of radar pulse Given the signal propagation speed, denoted by , is simply related to as: (1.1) As a result, the radar pulse contains the object range information H. -
Spectrum Estimation Using Periodogram, Bartlett and Welch
Spectrum estimation using Periodogram, Bartlett and Welch Guido Schuster Slides follow closely chapter 8 in the book „Statistical Digital Signal Processing and Modeling“ by Monson H. Hayes and most of the figures and formulas are taken from there 1 Introduction • We want to estimate the power spectral density of a wide- sense stationary random process • Recall that the power spectrum is the Fourier transform of the autocorrelation sequence • For an ergodic process the following holds 2 Introduction • The main problem of power spectrum estimation is – The data x(n) is always finite! • Two basic approaches – Nonparametric (Periodogram, Bartlett and Welch) • These are the most common ones and will be presented in the next pages – Parametric approaches • not discussed here since they are less common 3 Nonparametric methods • These are the most commonly used ones • x(n) is only measured between n=0,..,N-1 • Ensures that the values of x(n) that fall outside the interval [0,N-1] are excluded, where for negative values of k we use conjugate symmetry 4 Periodogram • Taking the Fourier transform of this autocorrelation estimate results in an estimate of the power spectrum, known as the Periodogram • This can also be directly expressed in terms of the data x(n) using the rectangular windowed function xN(n) 5 Periodogram of white noise 32 samples 6 Performance of the Periodogram • If N goes to infinity, does the Periodogram converge towards the power spectrum in the mean squared sense? • Necessary conditions – asymptotically unbiased: – variance -
Configuring Spectrogram Views
Chapter 5: Spectrographic Analysis Configuring spectrographic views To create a new spectrogram, spectrogram slice, or selection spectrum view, click on the appropriate New View button in the view toolbar (Figure 5.3) or choose a view type from the View > New menu. Figure 5.3. New View buttons Figure 5.3. The New View buttons, in Raven’s view toolbar. 112 Raven Pro 1.4 User’s Manual Chapter 5: Spectrographic Analysis A dialog box appears, containing parameters for configuring the requested type of spectrographic view (Figure 5.4). The dialog boxes for configuring spectrogram and spectrogram slice views are identical, except for their titles. The dialog boxes are identical because both view types calculate a spectrogram of the entire sound; the only difference between spectrogram and spectrogram slice views is in how the data are displayed (see “How the spectrographic views are related” on page 110). The dialog box for configuring a selection spectrum view is the same, except that it lacks the Averaging parameter. The remainder of this section explains each of the parameters in the configuration dialog box. Figure 5.4 Configure Spectrogram dialog Figure 5.4. The Configure New Spectrogram dialog box. Window type Each data record is multiplied by a window function before its spectrum is calculated. Window functions are used to reduce the magnitude of spurious “sidelobe” energy that appears at frequencies flanking each analysis frequency in a spectrum. These sidelobes appear as a result of Raven Pro 1.4 User’s Manual 113 Chapter 5: Spectrographic Analysis analyzing a finite (truncated) portion of a signal. -
Finite Impulse Response (FIR) Digital Filters (II) Ideal Impulse Response Design Examples Yogananda Isukapalli
Finite Impulse Response (FIR) Digital Filters (II) Ideal Impulse Response Design Examples Yogananda Isukapalli 1 • FIR Filter Design Problem Given H(z) or H(ejw), find filter coefficients {b0, b1, b2, ….. bN-1} which are equal to {h0, h1, h2, ….hN-1} in the case of FIR filters. 1 z-1 z-1 z-1 z-1 x[n] h0 h1 h2 h3 hN-2 hN-1 1 1 1 1 1 y[n] Consider a general (infinite impulse response) definition: ¥ H (z) = å h[n] z-n n=-¥ 2 From complex variable theory, the inverse transform is: 1 n -1 h[n] = ò H (z)z dz 2pj C Where C is a counterclockwise closed contour in the region of convergence of H(z) and encircling the origin of the z-plane • Evaluating H(z) on the unit circle ( z = ejw ) : ¥ H (e jw ) = åh[n]e- jnw n=-¥ 1 p h[n] = ò H (e jw )e jnwdw where dz = jejw dw 2p -p 3 • Design of an ideal low pass FIR digital filter H(ejw) K -2p -p -wc 0 wc p 2p w Find ideal low pass impulse response {h[n]} 1 p h [n] = H (e jw )e jnwdw LP ò 2p -p 1 wc = Ke jnwdw 2p ò -wc Hence K h [n] = sin(nw ) n = 0, ±1, ±2, …. ±¥ LP np c 4 Let K = 1, wc = p/4, n = 0, ±1, …, ±10 The impulse response coefficients are n = 0, h[n] = 0.25 n = ±4, h[n] = 0 = ±1, = 0.225 = ±5, = -0.043 = ±2, = 0.159 = ±6, = -0.053 = ±3, = 0.075 = ±7, = -0.032 n = ±8, h[n] = 0 = ±9, = 0.025 = ±10, = 0.032 5 Non Causal FIR Impulse Response We can make it causal if we shift hLP[n] by 10 units to the right: K h [n] = sin((n -10)w ) LP (n -10)p c n = 0, 1, 2, …. -
Signals and Systems Lecture 8: Finite Impulse Response Filters
Signals and Systems Lecture 8: Finite Impulse Response Filters Dr. Guillaume Ducard Fall 2018 based on materials from: Prof. Dr. Raffaello D’Andrea Institute for Dynamic Systems and Control ETH Zurich, Switzerland G. Ducard 1 / 46 Outline 1 Finite Impulse Response Filters Definition General Properties 2 Moving Average (MA) Filter MA filter as a simple low-pass filter Fast MA filter implementation Weighted Moving Average Filter 3 Non-Causal Moving Average Filter Non-Causal Moving Average Filter Non-Causal Weighted Moving Average Filter 4 Important Considerations Phase is Important Differentiation using FIR Filters Frequency-domain observations Higher derivatives G. Ducard 2 / 46 Finite Impulse Response Filters Moving Average (MA) Filter Definition Non-Causal Moving Average Filter General Properties Important Considerations Outline 1 Finite Impulse Response Filters Definition General Properties 2 Moving Average (MA) Filter MA filter as a simple low-pass filter Fast MA filter implementation Weighted Moving Average Filter 3 Non-Causal Moving Average Filter Non-Causal Moving Average Filter Non-Causal Weighted Moving Average Filter 4 Important Considerations Phase is Important Differentiation using FIR Filters Frequency-domain observations Higher derivatives G. Ducard 3 / 46 Finite Impulse Response Filters Moving Average (MA) Filter Definition Non-Causal Moving Average Filter General Properties Important Considerations FIR filters : definition The class of causal, LTI finite impulse response (FIR) filters can be captured by the difference equation M−1 y[n]= bku[n − k], Xk=0 where 1 M is the number of filter coefficients (also known as filter length), 2 M − 1 is often referred to as the filter order, 3 and bk ∈ R are the filter coefficients that describe the dependence on current and previous inputs. -
An Information Theoretic Algorithm for Finding Periodicities in Stellar Light Curves Pablo Huijse, Student Member, IEEE, Pablo A
IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 1, NO. 1, JANUARY 2012 1 An Information Theoretic Algorithm for Finding Periodicities in Stellar Light Curves Pablo Huijse, Student Member, IEEE, Pablo A. Estevez*,´ Senior Member, IEEE, Pavlos Protopapas, Pablo Zegers, Senior Member, IEEE, and Jose´ C. Pr´ıncipe, Fellow Member, IEEE Abstract—We propose a new information theoretic metric aligned to Earth. Periodic drops in brightness are observed for finding periodicities in stellar light curves. Light curves due to the mutual eclipses between the components of the are astronomical time series of brightness over time, and are system. Although most stars have at least some variation in characterized as being noisy and unevenly sampled. The proposed metric combines correntropy (generalized correlation) with a luminosity, current ground based survey estimations indicate periodic kernel to measure similarity among samples separated that 3% of the stars varying more than the sensitivity of the by a given period. The new metric provides a periodogram, called instruments and ∼1% are periodic [2]. Correntropy Kernelized Periodogram (CKP), whose peaks are associated with the fundamental frequencies present in the data. Detecting periodicity and estimating the period of stars is The CKP does not require any resampling, slotting or folding of high importance in astronomy. The period is a key feature scheme as it is computed directly from the available samples. for classifying variable stars [3], [4]; and estimating other CKP is the main part of a fully-automated pipeline for periodic light curve discrimination to be used in astronomical survey parameters such as mass and distance to Earth [5]. -
Windowing Design and Performance Assessment for Mitigation of Spectrum Leakage
E3S Web of Conferences 94, 03001 (2019) https://doi.org/10.1051/e3sconf/20199403001 ISGNSS 2018 Windowing Design and Performance Assessment for Mitigation of Spectrum Leakage Dah-Jing Jwo 1,*, I-Hua Wu 2 and Yi Chang1 1 Department of Communications, Navigation and Control Engineering, National Taiwan Ocean University 2 Pei-Ning Rd., Keelung 202, Taiwan 2 Bison Electronics, Inc., Neihu District, Taipei 114, Taiwan Abstract. This paper investigates the windowing design and performance assessment for mitigation of spectral leakage. A pretreatment method to reduce the spectral leakage is developed. In addition to selecting appropriate window functions, the Welch method is introduced. Windowing is implemented by multiplying the input signal with a windowing function. The periodogram technique based on Welch method is capable of providing good resolution if data length samples are selected optimally. Windowing amplitude modulates the input signal so that the spectral leakage is evened out. Thus, windowing reduces the amplitude of the samples at the beginning and end of the window, altering leakage. The influence of various window functions on the Fourier transform spectrum of the signals was discussed, and the characteristics and functions of various window functions were explained. In addition, we compared the differences in the influence of different data lengths on spectral resolution and noise levels caused by the traditional power spectrum estimation and various window-function-based Welch power spectrum estimations. 1 Introduction knowledge. Due to computer capacity and limitations on processing time, only limited time samples can actually The spectrum analysis [1-4] is a critical concern in many be processed during spectrum analysis and data military and civilian applications, such as performance processing of signals, i.e., the original signals need to be test of electric power systems and communication cut off, which induces leakage errors. -
Transformations for FIR and IIR Filters' Design
S S symmetry Article Transformations for FIR and IIR Filters’ Design V. N. Stavrou 1,*, I. G. Tsoulos 2 and Nikos E. Mastorakis 1,3 1 Hellenic Naval Academy, Department of Computer Science, Military Institutions of University Education, 18539 Piraeus, Greece 2 Department of Informatics and Telecommunications, University of Ioannina, 47150 Kostaki Artas, Greece; [email protected] 3 Department of Industrial Engineering, Technical University of Sofia, Bulevard Sveti Kliment Ohridski 8, 1000 Sofia, Bulgaria; mastor@tu-sofia.bg * Correspondence: [email protected] Abstract: In this paper, the transfer functions related to one-dimensional (1-D) and two-dimensional (2-D) filters have been theoretically and numerically investigated. The finite impulse response (FIR), as well as the infinite impulse response (IIR) are the main 2-D filters which have been investigated. More specifically, methods like the Windows method, the bilinear transformation method, the design of 2-D filters from appropriate 1-D functions and the design of 2-D filters using optimization techniques have been presented. Keywords: FIR filters; IIR filters; recursive filters; non-recursive filters; digital filters; constrained optimization; transfer functions Citation: Stavrou, V.N.; Tsoulos, I.G.; 1. Introduction Mastorakis, N.E. Transformations for There are two types of digital filters: the Finite Impulse Response (FIR) filters or Non- FIR and IIR Filters’ Design. Symmetry Recursive filters and the Infinite Impulse Response (IIR) filters or Recursive filters [1–5]. 2021, 13, 533. https://doi.org/ In the non-recursive filter structures the output depends only on the input, and in the 10.3390/sym13040533 recursive filter structures the output depends both on the input and on the previous outputs. -
Discrete-Time Fourier Transform
.@.$$-@@4&@$QEG'FC1. CHAPTER7 Discrete-Time Fourier Transform In Chapter 3 and Appendix C, we showed that interesting continuous-time waveforms x(t) can be synthesized by summing sinusoids, or complex exponential signals, having different frequencies fk and complex amplitudes ak. We also introduced the concept of the spectrum of a signal as the collection of information about the frequencies and corresponding complex amplitudes {fk,ak} of the complex exponential signals, and found it convenient to display the spectrum as a plot of spectrum lines versus frequency, each labeled with amplitude and phase. This spectrum plot is a frequency-domain representation that tells us at a glance “how much of each frequency is present in the signal.” In Chapter 4, we extended the spectrum concept from continuous-time signals x(t) to discrete-time signals x[n] obtained by sampling x(t). In the discrete-time case, the line spectrum is plotted as a function of normalized frequency ωˆ . In Chapter 6, we developed the frequency response H(ejωˆ ) which is the frequency-domain representation of an FIR filter. Since an FIR filter can also be characterized in the time domain by its impulse response signal h[n], it is not hard to imagine that the frequency response is the frequency-domain representation,orspectrum, of the sequence h[n]. 236 .@.$$-@@4&@$QEG'FC1. 7-1 DTFT: FOURIER TRANSFORM FOR DISCRETE-TIME SIGNALS 237 In this chapter, we take the next step by developing the discrete-time Fouriertransform (DTFT). The DTFT is a frequency-domain representation for a wide range of both finite- and infinite-length discrete-time signals x[n]. -
Understanding the Lomb-Scargle Periodogram
Understanding the Lomb-Scargle Periodogram Jacob T. VanderPlas1 ABSTRACT The Lomb-Scargle periodogram is a well-known algorithm for detecting and char- acterizing periodic signals in unevenly-sampled data. This paper presents a conceptual introduction to the Lomb-Scargle periodogram and important practical considerations for its use. Rather than a rigorous mathematical treatment, the goal of this paper is to build intuition about what assumptions are implicit in the use of the Lomb-Scargle periodogram and related estimators of periodicity, so as to motivate important practical considerations required in its proper application and interpretation. Subject headings: methods: data analysis | methods: statistical 1. Introduction The Lomb-Scargle periodogram (Lomb 1976; Scargle 1982) is a well-known algorithm for de- tecting and characterizing periodicity in unevenly-sampled time-series, and has seen particularly wide use within the astronomy community. As an example of a typical application of this method, consider the data shown in Figure 1: this is an irregularly-sampled timeseries showing a single ob- ject from the LINEAR survey (Sesar et al. 2011; Palaversa et al. 2013), with un-filtered magnitude measured 280 times over the course of five and a half years. By eye, it is clear that the brightness of the object varies in time with a range spanning approximately 0.8 magnitudes, but what is not immediately clear is that this variation is periodic in time. The Lomb-Scargle periodogram is a method that allows efficient computation of a Fourier-like power spectrum estimator from such unevenly-sampled data, resulting in an intuitive means of determining the period of oscillation. -
Effect of Data Gaps: Comparison of Different Spectral Analysis Methods
Ann. Geophys., 34, 437–449, 2016 www.ann-geophys.net/34/437/2016/ doi:10.5194/angeo-34-437-2016 © Author(s) 2016. CC Attribution 3.0 License. Effect of data gaps: comparison of different spectral analysis methods Costel Munteanu1,2,3, Catalin Negrea1,4,5, Marius Echim1,6, and Kalevi Mursula2 1Institute of Space Science, Magurele, Romania 2Astronomy and Space Physics Research Unit, University of Oulu, Oulu, Finland 3Department of Physics, University of Bucharest, Magurele, Romania 4Cooperative Institute for Research in Environmental Sciences, Univ. of Colorado, Boulder, Colorado, USA 5Space Weather Prediction Center, NOAA, Boulder, Colorado, USA 6Belgian Institute of Space Aeronomy, Brussels, Belgium Correspondence to: Costel Munteanu ([email protected]) Received: 31 August 2015 – Revised: 14 March 2016 – Accepted: 28 March 2016 – Published: 19 April 2016 Abstract. In this paper we investigate quantitatively the ef- 1 Introduction fect of data gaps for four methods of estimating the ampli- tude spectrum of a time series: fast Fourier transform (FFT), Spectral analysis is a widely used tool in data analysis and discrete Fourier transform (DFT), Z transform (ZTR) and processing in most fields of science. The technique became the Lomb–Scargle algorithm (LST). We devise two tests: the very popular with the introduction of the fast Fourier trans- single-large-gap test, which can probe the effect of a sin- form algorithm which allowed for an extremely rapid com- gle data gap of varying size and the multiple-small-gaps test, putation of the Fourier Transform. In the absence of modern used to study the effect of numerous small gaps of variable supercomputers, this was not just useful, but also the only size distributed within the time series.