This item is in: Engineering > Computing and electrical engineering
Digital filters and signal processing in electronic engineering: Theory, applications, architecture, codeS M Bozic and R J Chance, University of Birmingham, UK
Woodhead Publishing Series in Electronic and Optical Materials No. 4
- provides an unusual blend of theory and practice of digital signal processing (DSP)
- discusses fundamental signal processing procedures, convolution, correlation, the Discrete Fourier Transform and its fast computation algorithms
- includes number representations, multiply-accumulate, special addressing modes, zero overhead iteration schemes, and single and multiple instructions
From industrial and teaching experience the authors provide a blend of theory and practice of digital signal processing (DSP) for advanced undergraduate and post-graduate engineers reading electronics. This fast-moving, developing area is driven by the information technology revolution. It is a source book in research and development for embedded system design engineers, designers in real-time computing, and applied mathematicians who apph DSP techniques in telecommunications, aerospace (control systems), satellite communications, instrumentation, and medical technology (ultrasound and magnetic resonance imaging).
The book is particularly useful at the hardware end of DSP, with its emphasis on practical I)SP devices and the integration of basic processes with appropriate software. It is unique to find in one volume the implementation of the equations as algorithms, not only in \IATLAB but right up to a working DSP-based scheme. Other relevant architectural features include number representations, multiply-accumulate, special addressing modes, zero overhead iteration schemes. and single and multiple nlicroprocessors which will allow the readers to compare and understand both current processors and future DSP developments.
Fundamental signal processing procedures are introduced and developed: also convolution. correlation, the Discrete Fourier Transform and its fast computation algorithms. Then follo finite impulse response (FIR) filters, infinite impulse response (IlR) filters, multirate filters, adaptive filters, and topics from communication and control. I)esign examples are given in all of these cases, taken through an algorithm testing stage using MATLAB. The design of the latter. using C language models, is explained together with the experimental results of real time integer implementations.
Academic prerequisites are first and second year university mathematics, an introductor knowledge of circuit theor ‘and microprocessors. and C Language.
ISBN 1 898563 58 6
ISBN-13: 978 1 898563 58 7
October 1998
256 pages 244 x 172mm hardback
£50.00 / US$85.00 / €60.00

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Titles which may also be of interest:
Object-oriented technology and computing systems re-engineering
Circuit analysis
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Contents
Basic concepts and analytical tools
- Introduction
- Sampling and signal reconstruction
- ABC quantization noise
- The Z-transform
- Inverse Z-transform
- DSP Signal description
- Multirate digital signal processing
- Appendix 1.1: Frequency spectrum, form 1
- Appendix 1.2: Frequency spectrum form 1
- Appendix 1.3: Dirac function; Fourier pairs
- Appendix 1.4: Convolution equation derivation
Fundamental features of DSP
- Introduction
- DSP history and performance
- Number representation in DSPs
- The multiply-accumulate operation
- Fundamental DSP addressing modes
- Special DSP indirect addressing modes
- Special DSP iteration schemes
- Summary
- Appendix 2.1: TMS32OC2S/C50 floating point routines
Practical DSP devices
- Introduction
- Overview of commercial DSPs
- Motorola DSPS6000 &. DSP96000
- TMS320C25 and TMS32OC
- ADSP2100 and ADSP2IO2O
- DSP32 and TMS32OC3O
Discrete Fourier transform
- Introduction
- Discrete Fourier transform (DFT)
- Use of the DFT in convolution
- Use of the DFT in correlation
- Calculation of the DFT: FFT algorithms
- The Goertzel algorithm
- The Chirp Z-transform (CZT)
- Spectral analysis with computers
DSP implementations of Fourier and Goertzel algorithms
- Introduction
- An intuitive view of the FF1
- The FFT algorithm
- Some experiment.s with DSP FFT programs
- Goertzel implementation on a computer
- Goertzel and FFT results
- Appendix 5,1: TMS320C25 Assembler and instruction summary
- Appendix 5.2: 256 point radix 2 FFT for the TMS320C25/C50
- Appendix 5.3: Goertzel’s algorithm fhr the TMS320C25/C50
FIR filter design methods
- Introduction
- Special features of FIR filters
- Design based on Fourier series and windows
- Design based on frequency response sampling
- FIR filter design examples
- The implementation of decimation and interpolation
- Appendix 6.1: fMS320C25/C50 multirate FIR program
IIR filter design
- Introduction
- Impulse invariance method (II)
- The bilinear transform (BT) method
- Frequency transformations
- Design of analogue Iowpass filter prototypes
- IIR structures and implementations
- Quantization effects
- IIR filter design
- IIR filter implementation
- Appendix 7.1: Coefficient quantization
- Appendix 72: HR LP filter listing for the TMS320C25
- Appendix 7.3: HP TMS320C25 hR filter response
Other topics in digital signal processing
- Introduction
- Adaptive filters
- Deconvolution and system identification
- Homomorphic deconvolution
- Linear prediction
- Control system theory and design
- DSP implementation of adaptive filter and control
- Appendix 8 1 Table of Z-transforms and S-transforms
- Appendix 8.2 LMS adaptive filter listing for the TMS320C25
- Appendix 8.3 Digital control listing for the TMS320C25
References
