From the Publisher:
This is undoubtedly the most accessible book on digital signal processing (DSP) available to the beginner. Using intuitive explanations and well-chosen examples, this book gives you the tools to develop a fundamental understanding of DSP theory. The author covers the essential mathematics by explaining the meaning and significance of the key DSP equations. Comprehensive in scope, and gentle in approach, the book will help you achieve a thorough grasp of the basics and move gradually to more sophisticated DSP concepts and applications.

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Understanding digital signal processing

A location reporting paging communication system comprising space satellites, ground stations and a remote receiving unit adapted to resolve a global position from signals transmitted from a communication transmitter. The subscriber in possession of the remote receiving unit updates the paging network with global positioning information. A caller paging a subscriber in possession of the remote receiving unit may request the global location of the remote receiving unit. The paging network could divulge or block such information from a caller depending on the requirements of the subscriber.

Location reporting satellite paging system with optional blocking of location reporting

The basic concepts and building blocks in multirate digital signal processing (DSP), including the digital polyphase representation, are reviewed. Recent progress, as reported by several authors in this area, is discussed. Several applications are described, including subband coding of waveforms, voice privacy systems, integral and fractional sampling rate conversion (such as in digital audio), digital crossover networks, and multirate coding of narrowband filter coefficients. The M-band quadrature mirror filter (QMF) bank is discussed in considerable detail, including an analysis of various errors and imperfections. Recent techniques for perfect signal reconstruction in such systems are reviewed. The connection between QMF banks and other related topics, such as block digital filtering and periodically time-varying systems, is examined in a pseudo-circulant-matrix framework. Unconventional applications of the polyphase concept are discussed. >

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Multirate digital filters, filter banks, polyphase networks, and applications: a tutorial

A fractional delay filter is a device for bandlimited interpolation between samples. It finds applications in numerous fields of signal processing, including communications, array processing, speech processing, and music technology. We present a comprehensive review of FIR and allpass filter design techniques for bandlimited approximation of a fractional digital delay. Emphasis is on simple and efficient methods that are well suited for fast coefficient update or continuous control of the delay value. Various new approaches are proposed and several examples are provided to illustrate the performance of the methods. We also discuss the implementation complexity of the algorithms. We focus on four applications where fractional delay filters are needed: synchronization of digital modems, incommensurate sampling rate conversion, high-resolution pitch prediction, and sound synthesis of musical instruments.

Splitting the unit delay [FIR/all pass filters design]

Wave digital filters (WDFs) are modeled after classical filters, preferably in lattice or ladder configurations or generalizations thereof. They have very good properties concerning coefficient accuracy requirements, dynamic range, and especially all aspects of stability under finite-arithmetic conditions. A detailed review of WDF theory is given. For this several goals are set: to offer an introduction for those not familiar with the subject, to stress practical aspects in order to serve as a guide for those wanting to design or apply WDFs, and to give insight into the broad range of aspects of WDF theory and its many relationships with other areas, especially in the signal-processing field. Correspondingly, mathematical analyses are included only if necessary for gaining essential insight, while for all details of more special nature reference is made to existing literature.

Wave digital filters: Theory and practice

The digital filtering process can be achieved by a set of phase shifters with suitable characteristics. A particular set, named polyphase network, is defined and analyzed. It permits the use of recursive devices for efficient sample-rate alteration. The comparison with conventional filters shows that, with the same active memory, a reduction of computation rate approaching a factor of 2 can be achieved when the alteration factor increases. A more substantial gain can be obtained in the direct realization of a uniform bank of recursive filters through combination of the polyphase network with a discrete Fourier transform (DFT) computer; savings in hardware also result from the low sensitivity of the structure to coefficient word lengths.

Digital filtering by polyphase network:Application to sample-rate alteration and filter banks

Techniques applied to the digital processing of speech and signaling in a 60 channel FDM to 30 channel PCM converter are presented. Great efficiency is achieved through connecting a particular type of Fourier Transform computer to a polyphase network in which coefficient symmetries are exploited. Digital processing parameter values have been chosen so as to meet specified performance objectives. Realization of a laboratory model is outlined and measured performance is reported. Adequacy of the various options retained for the digital processing is confirmed by the experimental results.

M. Coudreuse

Papers

Digital filtering by polyphase network:Application to sample-rate alteration and filter banks

Digital Processing Techniques in the 60 Channel Transmultiplexer