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Analog and Digital Filters Design and Realization by Harry Y. F. Lam (z-lib.org).pdf
In signal processing , a digital filter is a system that performs mathematical operations on a sampled , discrete-time signal to reduce or enhance certain aspects of that signal.
This is in contrast to the other major type of electronic filter , the analog filter , which is an electronic circuit operating on continuous-time analog signals. A digital filter system usually consists of an analog-to-digital converter ADC to sample the input signal, followed by a microprocessor and some peripheral components such as memory to store data and filter coefficients etc.
Program Instructions software running on the microprocessor implement the digital filter by performing the necessary mathematical operations on the numbers received from the ADC. In some high performance applications, an FPGA or ASIC is used instead of a general purpose microprocessor, or a specialized digital signal processor DSP with specific paralleled architecture for expediting operations such as filtering.
Digital filters may be more expensive than an equivalent analog filter due to their increased complexity, but they make practical many designs that are impractical or impossible as analog filters. Digital filters can often be made very high order, and are often finite impulse response filters which allows for linear phase response.
When used in the context of real-time analog systems, digital filters sometimes have problematic latency the difference in time between the input and the response due to the associated analog-to-digital and digital-to-analog conversions and anti-aliasing filters , or due to other delays in their implementation.
Digital filters are commonplace and an essential element of everyday electronics such as radios , cellphones , and AV receivers.
A digital filter is characterized by its transfer function , or equivalently, its difference equation. Mathematical analysis of the transfer function can describe how it will respond to any input. As such, designing a filter consists of developing specifications appropriate to the problem for example, a second-order low pass filter with a specific cut-off frequency , and then producing a transfer function which meets the specifications. The transfer function for a linear, time-invariant, digital filter can be expressed as a transfer function in the Z -domain ; if it is causal, then it has the form: .
This is the form for a recursive filter , which typically leads to an IIR infinite impulse response behaviour, but if the denominator is made equal to unity i. A variety of mathematical techniques may be employed to analyze the behaviour of a given digital filter.
Many of these analysis techniques may also be employed in designs, and often form the basis of a filter specification. Typically, one characterizes filters by calculating how they will respond to a simple input such as an impulse. One can then extend this information to compute the filter's response to more complex signals. The impulse response is a characterization of the filter's behaviour. Digital filters are typically considered in two categories: infinite impulse response IIR and finite impulse response FIR.
In the case of linear time-invariant FIR filters, the impulse response is exactly equal to the sequence of filter coefficients, and thus:. IIR filters on the other hand are recursive, with the output depending on both current and previous inputs as well as previous outputs. The general form of an IIR filter is thus:.
Plotting the impulse response will reveal how a filter will respond to a sudden, momentary disturbance. An IIR filter will always be recursive. While it is possible for a recursive filter to have a finite impulse response, non-recursive filters will always have a finite impulse response.
An example is the moving average MA filter, that can be implemented both recursively [ citation needed ] and non recursively.
In discrete-time systems, the digital filter is often implemented by converting the transfer function to a linear constant-coefficient difference equation LCCD via the Z-transform.
The discrete frequency-domain transfer function is written as the ratio of two polynomials. For example:. The resultant linear difference equation is:. Applying the filter to an input in this form is equivalent to a Direct Form I or II see below realization, depending on the exact order of evaluation.
In plain terms, for example, as used by a computer programmer implementing the above equation in code, it can be described as follows:. Although filters are easily understood and calculated, the practical challenges of their design and implementation are significant and are the subject of much advanced research. There are two categories of digital filter: the recursive filter and the nonrecursive filter. These are often referred to as infinite impulse response IIR filters and finite impulse response FIR filters, respectively.
After a filter is designed, it must be realized by developing a signal flow diagram that describes the filter in terms of operations on sample sequences. A given transfer function may be realized in many ways.
In the same way, all realizations may be seen as "factorizations" of the same transfer function, but different realizations will have different numerical properties. Specifically, some realizations are more efficient in terms of the number of operations or storage elements required for their implementation, and others provide advantages such as improved numerical stability and reduced round-off error.
Some structures are better for fixed-point arithmetic and others may be better for floating-point arithmetic. A straightforward approach for IIR filter realization is direct form I , where the difference equation is evaluated directly.
This form is practical for small filters, but may be inefficient and impractical numerically unstable for complex designs. The alternate direct form II only needs N delay units, where N is the order of the filter — potentially half as much as direct form I. This structure is obtained by reversing the order of the numerator and denominator sections of Direct Form I, since they are in fact two linear systems, and the commutativity property applies.
The disadvantage is that direct form II increases the possibility of arithmetic overflow for filters of high Q or resonance. A common strategy is to realize a higher-order greater than 2 digital filter as a cascaded series of second-order "biquadratric" or "biquad" sections  see digital biquad filter. The advantage of this strategy is that the coefficient range is limited. Cascading direct form II sections results in N delay elements for filters of order N. Digital filters are not subject to the component non-linearities that greatly complicate the design of analog filters.
Analog filters consist of imperfect electronic components, whose values are specified to a limit tolerance e. As the order of an analog filter increases, and thus its component count, the effect of variable component errors is greatly magnified.
In digital filters, the coefficient values are stored in computer memory, making them far more stable and predictable. Because the coefficients of digital filters are definite, they can be used to achieve much more complex and selective designs — specifically with digital filters, one can achieve a lower passband ripple, faster transition, and higher stopband attenuation than is practical with analog filters. Even if the design could be achieved using analog filters, the engineering cost of designing an equivalent digital filter would likely be much lower.
Furthermore, one can readily modify the coefficients of a digital filter to make an adaptive filter or a user-controllable parametric filter.
While these techniques are possible in an analog filter, they are again considerably more difficult. Digital filters can be used in the design of finite impulse response filters. Equivalent analog filters are often more complicated, as these require delay elements. Digital filters rely less on analog circuitry, potentially allowing for a better signal-to-noise ratio. A digital filter will introduce noise to a signal during analog low pass filtering, analog to digital conversion, digital to analog conversion and may introduce digital noise due to quantization.
With analog filters, every component is a source of thermal noise such as Johnson noise , so as the filter complexity grows, so does the noise. However, digital filters do introduce a higher fundamental latency to the system. In an analog filter, latency is often negligible; strictly speaking it is the time for an electrical signal to propagate through the filter circuit.
In digital systems, latency is introduced by delay elements in the digital signal path, and by analog-to-digital and digital-to-analog converters that enable the system to process analog signals. In very simple cases, it is more cost effective to use an analog filter. Introducing a digital filter requires considerable overhead circuitry, as previously discussed, including two low pass analog filters.
Another argument for analog filters is low power consumption. Analog filters require substantially less power and are therefore the only solution when power requirements are tight. When making an electrical circuit on a PCB it is generally easier to use a digital solution, because the processing units are highly optimized over the years.
Making the same circuit with analog components would take up a lot more space when using discrete components. A filter can be represented by a block diagram , which can then be used to derive a sample processing algorithm to implement the filter with hardware instructions. A filter may also be described as a difference equation , a collection of zeros and poles or an impulse response or step response. Some digital filters are based on the fast Fourier transform , a mathematical algorithm that quickly extracts the frequency spectrum of a signal, allowing the spectrum to be manipulated such as to create very high order band-pass filters before converting the modified spectrum back into a time-series signal with an inverse FFT operation.
These filters give O n log n computational costs whereas conventional digital filters tend to be O n 2. Another form of a digital filter is that of a state-space model. A well used state-space filter is the Kalman filter published by Rudolf Kalman in Traditional linear filters are usually based on attenuation.
Alternatively nonlinear filters can be designed, including energy transfer filters,  which allow the user to move energy in a designed way so that unwanted noise or effects can be moved to new frequency bands either lower or higher in frequency, spread over a range of frequencies, split, or focused.
Energy transfer filters complement traditional filter designs and introduce many more degrees of freedom in filter design.
Digital energy transfer filters are relatively easy to design and to implement and exploit nonlinear dynamics. From Wikipedia, the free encyclopedia. Filter used on discretely-sampled signals in signal processing. This article only describes one highly specialized aspect of its associated subject. Please help improve this article by adding more general information. The talk page may contain suggestions. April Main article: Filter design. This section needs expansion.
You can help by adding to it. July The Related Media Group. Retrieved 13 July Rabiner and C. Rader, Eds. Digital Signal Processing. Proceedings of the IEEE. Wiley, Categories : Digital signal processing Synthesiser modules Signal processing filter.
Variable digital filters are widely used in a number of applications of signal processing because of their capability of self-tuning frequency characteristics such as the cutoff frequency and the bandwidth. This chapter introduces recent advances on variable digital filters, focusing on the problems of design and realization, and application to adaptive filtering. In the topic on design and realization, we address two major approaches: one is the frequency transformation and the other is the multi-dimensional polynomial approximation of filter coefficients. Digital Systems. Digital filter is well known as one of the essential and fundamental components in signal processing devices. In addition, many signal processing applications such as digital audio equipment and telecommunication systems sometimes require simultaneous realization of digital filtering and real-time control of filter characteristics.
Recent Advances in Variable Digital Filters
Filter design is the process of designing a signal processing filter that satisfies a set of requirements, some of which are contradictory. The purpose is to find a realization of the filter that meets each of the requirements to a sufficient degree to make it useful. The filter design process can be described as an optimization problem where each requirement contributes to an error function that should be minimized. Certain parts of the design process can be automated, but normally an experienced electrical engineer is needed to get a good result.
Lars Wanhammar was born in Vansbro, Sweden, on August 19, He received the Tekn. His research interests include theory and design of communication and digital signal processing systems, particularly analog and digital filters and fast transforms, computational properties of digital signal processing algorithms, computer-aided design tools, and ULSI techniques. He is the author or co-author of several textbooks.
Documentation Help Center. The primary advantage of IIR filters over FIR filters is that they typically meet a given set of specifications with a much lower filter order than a corresponding FIR filter. This allows for a noncausal, zero-phase filtering approach via the filtfilt function , which eliminates the nonlinear phase distortion of an IIR filter. This toolbox provides functions to create all these types of classical IIR filters in both the analog and digital domains except Bessel, for which only the analog case is supported , and in lowpass, highpass, bandpass, and bandstop configurations. For most filter types, you can also find the lowest filter order that fits a given filter specification in terms of passband and stopband attenuation, and transition width s. The direct filter design function yulewalk finds a filter with magnitude response approximating a specified frequency-response function. This is one way to create a multiband bandpass filter.
Bell Telephone Laboratories, Inc. Formerly with University of California, Berkeley. Prentice-Hall series in electrical and computer engineering Includes bibliographical references and index. Electric filters. F5L26 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.
Unlike most books on filters, Analog and Digital Filter Design does not start from a position of mathematical complexity. It is written to show readers how to design effective and working electronic filters. The background information and equations from the first edition have been moved into an appendix to allow easier flow of the text while still providing the information for those who are interested. The addition of questions at the end of each chapter as well as electronic simulation tools has allowed for a more practical, user-friendly text. Steve works alongside design engineers throughout Europe to design circuits using components made by Intersil Inc, a US based manufacturer of CMOS ICs used for power supply controllers and for analogue signal processing.
Documentation Help Center. The example concentrates on lowpass filters but most of the results apply to other response types as well. This example focuses on the design of digital filters rather than on their applications. If you want to learn more about digital filter applications, see Practical Introduction to Digital Filtering. Because the impulse response required to implement the ideal lowpass filter is infinitely long, it is impossible to design an ideal FIR lowpass filter.