Development of a Digital Universal Filter Bank

Total Page:16

File Type:pdf, Size:1020Kb

Development of a Digital Universal Filter Bank UPTEC E16 005 Examensarbete 30 hp Oktober 2016 Development of a Digital Universal Filter Bank Mats Larsson Abstract Development of a Digital Universal Filter Bank Mats Larsson Teknisk- naturvetenskaplig fakultet UTH-enheten This is a master's thesis project, which is a part of the Master Programme in Electrical Engineering at Uppsala university. Besöksadress: Ångströmlaboratoriet Lägerhyddsvägen 1 When developing a product or performing measurements, it is Hus 4, Plan 0 sometimes necessary to remove some content of a signal. This might be due to an interfering source that has to be filtered out, or Postadress: that only a specific frequency interval is of interest. In such a Box 536 751 21 Uppsala case, it would be practical if a universal frequency selective filter was available and easy to use. Telefon: 018 – 471 30 03 In this thesis, a platform for implementing different frequency Telefax: selective digital filters is developed. Through a user interface, 018 – 471 30 00 parameters such as sampling frequency, filter order, type of filter and cutoff frequencies are set by the user. This provides a Hemsida: platform which is easy to configure in order to run one or http://www.teknat.uu.se/student multiple IIR or FIR filters in various constellations. By combining different filters, a wide variety of frequency responses can be obtained. A prototype is constructed, which allows the user to connect up to two input signals and retrieve up to two output signals. The filter bank is programmed in C and implemented in a 32-bit microcontroller, base on the ARM architecture. To get a reliable prototype, a printed circuit board is designed and manufactured. To protect the electronics from external stress, a cover is designed and 3D-printed. The filter design algorithms and the algorithm used when running the filter bank gives satisfying results. The capacity of the filter bank can be increased by refining the filtering algorithm. By improving the hardware, noise generated by the electronics can be decreased. The filter bank has a really good potential to become a very useful tool. Handledare: Ping Wu Ämnesgranskare: Tomas Olofsson Examinator: Mikael Bergkvist ISSN: 1654-7616, UPTEC E16 005 Sammanfattning Denna rapport ¨arett resultat fr˚anett examensarbete inom civilingenj¨orsprogrammet i elektroteknik vid Uppsala Universitet. Vid t.ex. utveckling av produkter eller inom m¨atteknikskulle det underl¨attaom det fanns m¨ojlighetatt p˚aett enkelt s¨attfiltrera bort delar av inneh˚alleti en signal. Detta d˚at.ex. en st¨orningfortplantat sig i ett m¨atsystem. De befintliga systemen som finns tillg¨angligap˚amarknaden kr¨aver oftast programmering i specifik mjukvara, via t.ex. en PC, och tar d¨arf¨ortid att implementera. Mjukvaran f¨ordessa system medf¨ordessutom oftast kostnader, d˚alicenser kr¨avs. I detta examensarbete utvecklas en plattform som erbjuder ett snabbt och enkelt s¨attatt implementera och kombinera olika digitala frekvensselektiva filter. Den universala digitala filterbanken kan hantera upp till tv˚ain- och utsignaler. Genom ett anv¨andargr¨anssnitt best˚aendeav en display och en knappsats, kan ett eller flera filter konfigueras f¨oratt sedan kopplas samman i filterbanken. P˚adetta s¨attkan anv¨andarenforma frekvenssvaret efter det behov som finns. Filterbanken ¨arprogrammerad i C och implementerad i en 32-bitars mikrokontroller. Den slutliga prototypen fungerar som ¨onskat och l˚ateranv¨andarenp˚aett mycket enkelt s¨attimplementera ett stort antal olika typer av frekvensselektiva filter. Anv¨andaren best¨ammer sj¨alvparrametrar s˚asomfiltrets ordningstal och brytfrekvenser. Olika filter kan dessutom kombineras genom att kopplas i serie eller parralellt, efter det behov som finns. Genom vidareutveckling av h˚ardvaran har filterbanken stor potential f¨oratt bli ett mycket anv¨andbartverktyg. Jag skulle vilja rikta ett tack till Ping Wu p˚aavdelningen f¨orsignaler och system vid Uppsala universitet som varit min handledare genom projektet samt Etteplan i V¨aster˚as som bist˚attmed tid f¨orkonsultation g¨allandeprogrammering. Jag skulle ocks˚avilja tacka Robin Hindgren f¨ordiskussioner och f¨orslag till f¨orb¨attringarav kretskortkonstruktionen samt Fredrik Evestedt p˚aavdelningen f¨orelectricitetsl¨aravid Uppsala universitet f¨or hj¨alpen med att skriva ut h¨oljettill prototypen. II Contents 1 Introduction 1 1.1 Background . .1 1.2 Aim of the project . .1 1.3 Project description . .1 1.4 Outline of the report . .4 2 Theory 5 2.1 Linear phase delay . .5 2.2 Sallen-Key topology . .6 2.3 Differential amplifier . .7 2.4 Analog to digital conversion . .8 2.5 Anti-aliasing filter . .9 2.6 Digital filter . 10 2.6.1 Finite impulse response filter . 11 2.6.2 Infinite impulse response filters . 19 2.7 Digital to analog conversion . 26 2.8 Reconstruction filter . 26 3 Hardware implementation 27 3.1 Design of analog signal chain . 27 3.1.1 DC-blocker . 27 3.1.2 DC-adaption . 28 3.1.3 Anti-aliasing and reconstruction filter . 28 3.1.4 Buffer . 28 3.1.5 Current limiter . 29 3.2 Microcontroller . 29 3.3 User interface . 30 3.4 Printed circuit board . 31 3.5 3D-printed cover . 32 4 Software implementation 34 4.1 Filter design algorithms . 35 4.2 Filter structure . 35 4.3 Output structure . 36 4.4 Filter bank algorithm . 37 4.5 User interface menu system . 38 5 Results 43 5.1 Hardware . 43 5.1.1 DC-blocker . 43 5.1.2 Anti-aliasing filter and reconstruction filters . 44 5.1.3 Prototype . 48 5.2 Software . 50 5.2.1 Filter validation . 50 III 5.2.2 User interface . 59 6 Discussion 64 7 Further development 66 Appendix A IIR filter coefficients 68 Appendix B Derivation of Sallen-Key topology 73 Appendix C Derivation of differential amplifier circuit 75 IV Glossary AAF Anti-Aliasing Filter. ADC Analog to Digital Converter. ARM Advanced RISC Machine. BP Band Pass. BS Band Stop. CPU Central Processing Unit. CT Continuous Time. DAC Digital to Analog Converter. DMA Direct Memory Access. DSP Digital Signal Processing. DT Discrete Time. FFT Fast Fourier Transform. FIR Finite Impulse Response. GPIO General-Purpose Input/Output. HP High Pass. IIR Infinite Impulse Response. KCL Kirchhoff's Current Law. LCD Liquid-Crystal Display. LP Low Pass. LTI Linear Time Invariant. MCU Microcontroller Unit. MIMO Multiple Input Multiple Output. MISO Multiple Input Single Output. PCB Printed Circuit Board. V RECF Reconstruction Filter. SIMO Single Input Multiple Output. SISO Single Input Single Output. VI 1 Introduction 1.1 Background Frequency selective filtering is a very important topic within the field of signal processing. By designing a filter it is possible to remove unwanted frequency components such as band-limited noise or an interfering signal. Digital filtering is a widely used method for filtering almost any signal, such as sensor signals and audio signals. In comparison to an analog filter, a digital filter is implemented in software, and therefore often no extra components need to be added. Also analog components characteristics tend to drift due to change of the ambient temperature, which is not the case for digital filters. Currently, there is no universal digital filter on the market, which allows the user to easily implement almost any filter without a sophisticated user interface, which is based on an additional system, such as a PC. 1.2 Aim of the project When developing a product, performing measurements on a system or any other situation requiring setting up a test rig, it would sometimes simplify the procedure if a specific filter was available and easy to implement. The aim of this project is to design such a universal filter bank where various filters can be designed and implemented to a system where one or multiple analog signals are available and have to be filtered. In this way, it is no longer necessary to bother about tedious and time-consuming calculations, it will also be easier to tune the filter for the specific application. 1.3 Project description In this thesis, a digital universal filter bank is designed and implemented in a Micro- controller Unit (MCU) based on the Advanced RISC Machine (ARM) architecture. The design allows the user to connect up to two input signals and achieve up to two output signals, where the maximum amplitude of the input signal is limited to 3.3 V peak-to- peak. Through a user interface, the user selects an appropriate filter for the given application and enter all the design parameters for the filter to be designed. When all design pa- rameters are selected, the filter bank automatically generates the filter. The filter bank is capable of running multiple filters simultaneously. In this way it is possible to connect different filters in series or in parallel to achieve the frequency response required for the given application. Since the MCU has a multiple channel Analog to Digital Converter (ADC) and Digital to Analog Converter (DAC), it is possible to configure the filter bank into a Single Input Single Output (SISO), Single Input Multiple Output (SIMO), Multiple Input Single Output (MISO) or a Multiple Input Multiple Output (MIMO) system. 1 The user interface allows the user to perform the following actions: • Select sampling frequency, either one of the three predefined sampling frequencies (16kHz, 48kHz or 96kHz), which are using an Anti-Aliasing Filter (AAF) and a Reconstruction Filter (RECF), or select a custom sampling frequency. • Add a new filter to the system. • Select Finite Impulse Response (FIR) or Infinite Impulse Response (IIR) filter. • Select an analog reference filter such as Butterworth or Chebyshev I if an IIR filter is designed, see section 2.6.2.1. • Select window function if a FIR filter is designed, see section 2.6.1.2. • Select filter order. • Select Low Pass (LP), High Pass (HP), Band Pass (BP) or Band Stop (BS) filter.
Recommended publications
  • Filter Banks, Time-Frequency Transforms and Applications
    Signal Analysis: Wavelets,Filter Banks, Time-Frequency Transforms and Applications. Alfred Mertins Copyright 0 1999 John Wiley& Sons Ltd print ISBN 0-471-98626-7 Electronic ISBN 0-470-84183-4 Chapter 6 Filter Banks Filter banks are arrangementsof low pass, bandpass, and highpass filters used for the spectral decomposition and composition of signals. They play an im- portant role in many modern signal processing applications such as audio and image coding. The reason for their popularity is the fact that they easily allow the extractionof spectral components of a signal while providing very efficient implementations. Since most filter banks involve various sampling rates, they are also referred to as multirate systems. To give an example,Figure 6.1 shows an M-channel filter bank. The input signal is decomposed into M so- called subb and signalsby applying M analysis filters with different passbands. Thus, each of the subband signals carries information on the input signal in a particular frequency band. The blocks with arrows pointing downwards in Figure 6.1 indicate downsampling (subsampling) by factor N, and the blocks with arrows pointing upwards indicate upsampling by N. Subsampling by N means that only every Nth sample is taken. This operation serves to reduce or eliminate redundancies in the M subband signals. Upsampling by N means the insertion of N - 1 consecutive zeros between the samples. This allows us to recover the original sampling rate. The upsamplers are followed by filters which replace the inserted zeros with meaningful values. In the case M = N we speak of critical subsampling, because this is the maximum downsampling factor for which perfect reconstruction can be achieved.
    [Show full text]
  • Chapter 2 – the Discrete Fourier Transform
    Ivan W. Selesnick et al. "The Discrete Fourier Transform" The Transform and Data Compression Handbook Ed. K. R. Rao et al. Boca Raton, CRC Press LLC, 2001 © 20001 CRC Press LLC Chapter 2 The Discrete Fourier Transform Ivan W. Selesnick Polytechnic University Gerald Schuller Bell Labs 2.1 Introduction The discrete Fourier transform (DFT) is a fundamental transform in digital signal processing, with applications in frequency analysis, fast convolution, image process- ing, etc. Moreover, fast algorithms exist that make it possible to compute the DFT very efficiently. The algorithms for the efficient computation of the DFT are collectively called fast Fourier transforms (FFTs). The historic paper by Cooley and Tukey [15] N N N made well known an FFT of complexity log2 , where is the length of the data vector. A sequence of early papers [3, 11, 13, 14, 15] still serves as a good reference for the DFT and FFT. In addition to texts on digital signal processing, a number of books devote special attention to the DFT and FFT [4, 7, 10, 20, 28, 33, 36, 39, 48]. The importance of Fourier analysis in general is put forth very well by Leon Co- hen [12]: . Bunsen and Kirchhoff, observed (around 1865) that light spectra can be used for recognition, detection, and classification of substances because they are unique to each substance. This idea, along with its extension to other waveforms and the invention of the tools needed to carry out spectral decomposition, certainly ranks as one of the most important discoveries in the history of mankind. The kth DFT coefficient of a length N sequence {x(n)} is defined as N−1 kn X(k) = x(n) WN ,k= 0,...,N − 1 (2.1) n=0 where π π W = e−j2π/N = 2 − j 2 N cos N sin N nk is the principal N-th root of unity.
    [Show full text]
  • Learning Sparsifying Filter Banks
    Learning Sparsifying Filter Banks Luke Pfister and Yoram Bresler Dept. of Electrical and Computer Engineering University of Illinois at Urbana-Champaign ABSTRACT Recent years have numerous algorithms to learn a sparse synthesis or analysis model from data. Recently, a generalized analysis model called the 'transform model' has been proposed. Data following the transform model is approximately sparsified when acted on by a linear operator called a sparsifying transform. While existing transform learning algorithms can learn a transform for any vectorized data, they are most often used to learn a model for overlapping image patches. However, these approaches do not exploit the redundant nature of this data and scale poorly with the dimensionality of the data and size of patches. We propose a new sparsifying transform learning framework where the transform acts on entire images rather than on patches. We illustrate the connection between existing patch-based transform learning approaches and the theory of block transforms, then develop a new transform learning framework where the transforms have the structure of an undecimated filter bank with short filters. Unlike previous work on transform learning, the filter length can be chosen independently of the number of filter bank channels. We apply our framework to accelerating magnetic resonance imaging. We simultaneously learn a sparsifying filter bank while reconstructing an image from undersampled Fourier measurements. Numerical experiments show our new model yields higher quality images than previous patch based sparsifying transform approaches. Keywords: Sparisfying transform learning, Sparse representations, Filter banks, MR reconstruction 1. INTRODUCTION Problems in fields ranging from statistical inference to medical imaging can be posed as the recovery of high- quality data from incomplete or corrupted linear measurements.
    [Show full text]
  • Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing Cancellation
    IEEE TRANSACTIONS ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOL.ASSP-34, NO. 5, OCTOBER 1986 1153 Analysis/Synthesis Filter Bank Design Basedon Time Domain Aliasing Cancellation JOHN P. PRINCEN, STUDENT MEMBER, IEEE, AND ALANBERNARD BRADLEY Abstract-A single-sideband analysis/synthesis system is proposed which provides perfect reconstruction aof signal froma set of critically sampled analysis signals. The technique is developed in terms of a dTr weighted overlap-add methodof analysis/synthesis and allows overlap q-1 x(n) %nl between adjacent time windows. This implies that time domain aliasing is introduced in the analysis; however, this aliasing is cancelledin the synthesis process, and the system can provide perfect reconstruction. k=O... K-1 Achieving perfect reconstruction places constraints on the time domain window shape which are equivalent to those placed on the frequency Fig. 1. Analysidsynthesis system framework. domain shape of analysis/synthesis channels used in recently proposed ily expressed in the frequency domain. Reconstruction can critically sampled systems based on frequency domain aliasing cancel- lation [7], [8]. In fact, a duality exists between the new technique and be obtained if the composite analysidsynthesis channel the frequency domain techniques of [7] and 181. The proposed tech- responses overlap and add such that their sum is flat in nique is more-efficient than frequency domain designs fora given num- the frequency domain. Any frequency domain aliasing in- ber of analysis/synthesis channels, and can provide reasonably band- troduced by representing the narrow-band analysis signals limited channel responses. The technique could be particularly useful at areduced sample rate mustbe removed in the synthesis in applicationswhere critically sampled analysislsynthesis is desir- able, e.g., coding.
    [Show full text]
  • FPGA-Based Filterbank Implementation for Parallel Digital Signal Processing 1
    5.].I FPGA-Based Filterbank Implementation for Parallel Digital Signal Processing 1 Stephan Berner and Phillip De Leon New Mexico State University Center for Space Telecommunications and Telemetering Box 30001, Dept. 3-0 Las Cruces, New Mexico 88003-8001 {sberner, pdeleon}_)nmsu.edu Abstract - One approach to parallel digital signal processing decomposes a high bandwidth signal into multiple lower bandwidth (rate) signals by an analysis bank. After processing, the subband signals are recombined into a fullband output signal by a synthesis bank. This paper describes an implementation of the analysis and synthesis banks using FPGAs. 1 Introduction In the last decade research in filter banks and subband processing as well as multi-resolution analysis (MRA) a.k.a, wavelet transforms has occurred at a fast pace [12],[14],[6].[3]. In particular, subband processing of signals is now common in a number of apl)lications such as audio, image, and video encoders/decoders; acoustic echo cancelers used in hands-free teleconferencing systems; and spread spectrum comnmnications systems [1]. In addition. emerging applications include multiple target tracking in radar, high-data rate modems for satellite channels, and suppression of interference signals in wireless applications [8].[13].[2]. As compared to equivalent fullband processing, subband signal processing in these applica- tions often leads to a performance increase due to the MRA and potentially a reduction in computation due to the lower sampling rate of the subband (component) signals. Another emerging area of application is in parallel digital signal processing architectures [5I. There are essentially two approaches in which to utilize multiple processors in an archi- tecture.
    [Show full text]
  • Lecture 44: Tutorial on Two-Band Filter Bank Prof.V.M.Gadre, EE, IIT Bombay
    WAVELETS AND MULTIRATE DIGITAL SIGNAL PROCESSING Lecture 44: Tutorial on Two-Band Filter Bank Prof.V.M.Gadre, EE, IIT Bombay 1 Two Band Filter Bank The 2-band filter bank have two section. The analysis section and the synthesis section. π H0(Z) is a low pass filter with a cut off frequency 2 and H1(Z) is a high pass filter with a cut π off frequency 2 . Analysis section analyzes or breaks down the input in two components. π G0(Z) is a low pass filter with a cut off frequency 2 and G1(Z) is a high pass filter with a cut π off frequency 2 . Synthesis section re-synthesize output from the inputs. It is impossible situation as we can never reach ideal low pass or high pass filter. Even so, it is possible to build perfect reconstruction structure. For e.g., if we take Haar 2-band filter bank, we have set of filters H0, H1, G0 and G1 all of them have impulse response of length 2 which can create perfect reconstruction situation i.e., output Y0 is same as input X0 except for a constant multiplier and a shift. 2 Haar 2-Band filter bank −1 H0(Z) = (1 + Z ) (1) 44 - 1 −1 H1(Z) = (−1 + Z ) (2) (1 + Z−1) G (Z) = (3) 0 2 (1 − Z−1) G (Z) = (4) 1 2 1 The factor of 2 can either be on the analysis or synthesis side. Let us take x[n] be the input to this Haar 2-band filter bank.
    [Show full text]
  • ˆ K  X(N)E N (1) S Xx  Directly from Signal Being Estimated X(N)
    International Journal of Modern Engineering Research (IJMER) www.ijmer.com Vol.1, Issue.2, pp-247-251 ISSN: 2249-6645 Performance Evaluation of Filter-bank based Spectrum Estimator M.Venakatanarayana1, Dr.T.Jayachandra Prasad 2 1Assoc . Prof , Dept. of ECE, KSRM College of Engg., Kadapa. 2Professor, Dept. of ECE, RGMCET, Nandyal, Abstract— signal. Estimator based on parametric method provides higher In this paper on attempt has been made to study the degree of detail. performance of Filter-bank based nonparametric spectral The disadvantage of parametric method is that if the signal is estimation. Several methods are available to estimate non not sufficiently and accurately described by the model, the parametric power spectrum. The band pass filter, which result is less meaningful. Non Parametric methods, on the sweeps through the frequency interval of interest, is main other hand, do not have any assumption about the shape of the element in power spectrum estimation setup. The filter- bank based spectrum estimation is developed and is power spectrum and try to find acceptable estimate of the applied to multi tone signal. The spectrum estimated based power spectrum without prior knowledge about the underlying on filter-bank approach has been compared with stochastic approach. The following sub-sections give review conventional nonparametric spectrum estimation on some of the spectrum estimation methods. techniques such as Periodogram, Welch and Blackman- Tukey. It is observed that the filter-bank method gives A. Periodogram better frequency resolution and low statistical variability. The most commonly known spectrum estimation technique is It is also found there is a tradeoff between resolution and periodogram, which is classified as a non parametric estimator.
    [Show full text]
  • Spectral Analysis of Signals
    \sm2" i i 2004/2/22 page i i i SPECTRAL ANALYSIS OF SIGNALS Petre Stoica and Randolph Moses PRENTICE HALL, Upper Saddle River, New Jersey 07458 i i i i \sm2" i i 2004/2/22 page ii i i Library of Congress Cataloging-in-Publication Data Spectral Analysis of Signals/Petre Stoica and Randolph Moses p. cm. Includes bibliographical references index. ISBN 0-13-113956-8 1. Spectral theory (Mathematics) I. Moses, Randolph II. Title 512'{dc21 2005 QA814.G27 00-055035 CIP Acquisitions Editor: Tom Robbins Editor-in-Chief: ? Assistant Vice President of Production and Manufacturing: ? Executive Managing Editor: ? Senior Managing Editor: ? Production Editor: ? Manufacturing Buyer: ? Manufacturing Manager: ? Marketing Manager: ? Marketing Assistant: ? Director of Marketing: ? Editorial Assistant: ? Art Director: ? Interior Designer: ? Cover Designer: ? Cover Photo: ? c 2005 by Prentice Hall, Inc. Upper Saddle River, New Jersey 07458 All rights reserved. No part of this book may be reproduced, in any form or by any means, without permission in writing from the publisher. Printed in the United States of America 10 9 8 7 6 5 4 3 2 1 ISBN 0-13-113956-8 Pearson Education LTD., London Pearson Education Australia PTY, Limited, Sydney Pearson Education Singapore, Pte. Ltd Pearson Education North Asia Ltd, Hong Kong Pearson Education Canada, Ltd., Toronto Pearson Educacion de Mexico, S.A. de C.V. Pearson Education - Japan, Tokyo Pearson Education Malaysia, Pte. Ltd i i i i \sm2" i i 2004/2/22 page iii i i Contents 1 Basic Concepts 1 1.1 Introduction . 1 1.2 Energy Spectral Density of Deterministic Signals .
    [Show full text]
  • Learning Filter Bank Sparsifying Transforms Luke Pfister, Student Member, IEEE, Yoram Bresler, Fellow, IEEE
    1 Learning Filter Bank Sparsifying Transforms Luke Pfister, Student Member, IEEE, Yoram Bresler, Fellow, IEEE Abstract Data is said to follow the transform (or analysis) sparsity model if it becomes sparse when acted on by a linear operator called a sparsifying transform. Several algorithms have been designed to learn such a transform directly from data, and data-adaptive sparsifying transforms have demonstrated excellent performance in signal restoration tasks. Sparsifying transforms are typically learned using small sub-regions of data called patches, but these algorithms often ignore redundant information shared between neighboring patches. We show that many existing transform and analysis sparse representations can be viewed as filter banks, thus linking the local properties of patch-based model to the global properties of a convolutional model. We propose a new transform learning framework where the sparsifying transform is an undecimated perfect reconstruction filter bank. Unlike previous transform learning algorithms, the filter length can be chosen independently of the number of filter bank channels. Numerical results indicate filter bank sparsifying transforms outperform existing patch-based transform learning for image denoising while benefiting from additional flexibility in the design process. Index Terms sparsifying transform, analysis model, analysis operator learning, sparse representations, perfect reconstruction, filter bank. I. INTRODUCTION Countless problems, from statistical inference to geological exploration, can be stated as the recovery of high-quality data from incomplete and/or corrupted linear measurements. Often, recovery is possible only if a model of the desired signal is used to regularize the recovery problem. A powerful example of such a signal model is the sparse representation, wherein the signal of interest admits a representation with few nonzero coefficients.
    [Show full text]
  • Digital Speech Processing— Lecture 10 Short-Time Fourier Analysis
    Digital Speech Processing— Lecture 10 Short-Time Fourier Analysis Methods - Filter Bank Design 1 Review of STFT ∞ jjmωωˆˆˆ − 1.()[][]Xenˆ =−∑ xmwnme m=−∞ function of nˆ (looks like a time sequence) function of ωˆ (looks like a transform) jωˆ ˆ Xenˆ ( ) defined for n=≤≤123, , ,...; 0 ωπˆ jωˆ 2. Interpretations of Xenˆ ( ) ˆ jω ˆ 1.n fixed, ωω==−ˆ variable; Xenˆ ( ) DTFT⎣⎡ xmwnm [ ] [ ]⎦⎤ ⇒⇒DFT View OLA implementation jjnωωˆˆ− 2. nn==∗ˆ variable, ωˆ fixed; Xen ( ) xne [ ] wn [ ] ⇒⇒Linear Filtering view filter bank implementation ⇒ FBS implementation 2 Review of STFT 3. Sampling Rates in Time and Frequency ⇒ ˆ jωˆ recover xn[ ] exactly from Xnˆ ( e ) 1. time: We(jω ) has bandwidth of B Hertz ⇒ 2B samples/sec rate 2F Hamming Window: B =⇒S (Hz) sample at L 4F S (Hz) or every L/4 samples L 2. frequency: wn[ ] is time limited to L samples ⇒ need at least L frequency samples to avoid time aliasing 3 Review of STFT with OLA method can recover xn( ) exactly using lower sampling rates in either time or frequency, e.g., can sample every L samples (and divide by window), or can use fewer than L frequency samples (filter bank channels); but these methods are highly subject to aliasing errors with any modifications to STFT can use windows (LPF) that are longer than L samples and still recover with NL<⇒ frequency channels; e.g., ideal LPF is infinite in time duration, but with zeros spaced N samples apart where FNS / is the BW of the ideal LPF 4 Review of STFT jω H0(e ) jω H1(e ) X(ejω) Y(ejω) + … jω HN-1(e ) jnωk hnk []= wne [] N−1 jjωω He%()==∑ Hk () e 1 k =0 N−1 hn%[]===∑ hk [] nδ [] n wnpn [][] k =0 ∞ pn[]=− N∑ δ [ n rN ] r =−∞ ∞ hn%[]=−=++ N∑ wrN [ ][δ n rN ] w []0 wN [ ] ..
    [Show full text]
  • Classical Sampling Theorems in the Context of Multirate and Polyphase Digital Filter Bank Structures
    1480 IEEE TRANSACTIONS ON ACOUSTICS. SPEECH, AND SIGNAL PROCESSING, VOL. 36, NO. 9, SEPTEMBER 1988 Classical Sampling Theorems in the Context of Multirate and Polyphase Digital Filter Bank Structures Abstract-The sampling theorem has been generalized in several di- frequency is at least equal to the Nyquist frequency 8 rections since its introduction during the first half of this century. Some 2a. One of the earliest extensions of this theorem was of these include derivative sampling theorems and nonuniform sam- pling theorems. Both of these have also been reinterpreted in terms of stated by Shannon himself in his 1949 paper [l], which a multichannel sampling framework. In the world of digital signal pro- says that if x, (t)and its first M - 1 derivatives are avail- cessing, multirate systems have gained substantial attention during the able, then uniformly spaced samples of these, taken at the last 1: decades. A system of this type is the maximally decimated anal- reduced rate of 8/M, are sufficient to reconstruct x,(t). ysislsynthesis filter bank, widely used in subband coding, voice privacy This result will be referred to as the derivative sampling systems, and spectral analysis. Some of the generalizations of the sam- pling theorem can be understood by relating the sampling problem to theorem in this paper. A different type of extension called the continuous-time version of the analysis/synthesis filter bank system, the nonuniform sampling theorem, which was stated by as was done by Papoulis and Brown. Basically, the recovery of a signal Black [7, p. 411 (and attributed to Cauchy, 1841!), has from “generalized samples” is a problem of designing appropriate lin- also been reviewed recently [4]-[6] in various contexts.
    [Show full text]
  • Two-Channel Filter Banks and Dyadic Decompositions
    Two-Channel Filter Banks and Dyadic Decompositions by D.S.G. Pollock University of Leicester Email: stephen [email protected] In the basic subband coding procedure, which has been used in the transmission of speech via digital media, the signal is first split between two frequency bands via complimentary highpass and lowpass filters and then it is subsampled and encoded for transmission. At the receiving end, the signals of the two channels are decoded and then upsampled, by the interpolation of zeros between the sample elements. Then, they are smoothed by filtering, before they are recom- bined in order to provide a representation the original signal. The objective here is a perfect reconstruction of the original signal. In a wavelets analysis, there is a further requirement, which is that the basis functions or wavelets, in terms of which the continuous signal may be expressed, should be mutually orthogonal both sequentially, which is within the channels, and laterally, which is across the channels. The orthogonality of the wavelets is guaranteed if the corresponding filters, which comprise the coefficients of the wavelets dilation equations, are mutually orthogonal in both ways. In this chapter, we shall derive the conditions for perfect reconstruction, which place certain restrictions on the filter coefficients. Then, we shall proceed to derive the more stringent conditions of orthogonality. First, we must consider the basic procedures for dividing the signal into a high-frequency and a low- frequency component and for transmitting and recombining these components. Once the architecture of the two-channel filter bank has been established, it can be used in pursuit of a dyadic decomposition of the frequency range in which, in descending the frequency scale, successive frequency bands have half the width of their predecessors.
    [Show full text]