scispace - formally typeset
Search or ask a question

Showing papers on "Digital signal processing published in 1977"


Book
01 Jan 1977
TL;DR: In this chapter,sequency as a generalized frequency is introduced, and the frequency is used as a parameter to distinguish individual functions that belong to sets of nonsinusoidal functions.
Abstract: sequency as a generalized frequency is introduced, and the frequency is used as a parameter to distinguish individual functions that belong to sets of nonsinusoidal functions. The sixth chapter is devoted to the study of the Walsh-Hadamard transform (WHT) and algorithms to compute it. The concept of the Walsh spectra and their properties are presented with physical significance. Special attention is given to the analogy between the Walsh-Hadamard and the discrete Fourier transforms. In Chapter 7, a study is made of the generalized Haar, Slant, and discrete cosine transforms. Fast algorithms to compute these transforms

2,372 citations


Book
01 Jan 1977
TL;DR: This chapter introduces the z-Transform, a new type of transform that combines Laplace Transforms, z-Transforms and Modified Z-TRANSFORMS with Convolution Integral to achieve state-of-the-art control of Discrete-Data Systems.
Abstract: Each chapter ends with Problems and References Chapter One includes only References) Introduction 1. Signal Conversion and Processing 2. The z-Transform 3. Transfer Functions, Block Diagrams and Signal Flow Graphs 4. The State Cariable Technique 5. Controllability, Observability and Stability 6. Frequency-Domain Analysis 7. Digital Simulation and Digital Redesign 8. Design of Discrete-Data Systems 9. Optimal Control 10. Micro-processor and DSP Controls Appendices: A. Fixed-Point and Floating-Point Numbers B. Mathematical Modeling of Sampling by Convolution Integral C. Table of Laplace Transforms, z-Transforms and Modified Z-TRANSFORMS D. General Gain Formula for Signal Flow Graphs E. Routh's Tabulation for Stability Analysis F. Galil DMC-100 Motion Controller Board

1,070 citations


Journal ArticleDOI
D.E. Dudgeon1
01 Jun 1977
TL;DR: The purpose of this paper is mainly tutorial, to describe mathematically and intuitively the fundamental relationships necessary to understand digital array processing.
Abstract: With the advent of high-speed digital electronics, it has become feasible to use digital computers and special purpose digital processors to perform the computational tasks associated with signal reception using an antenna or directional array. The purpose of this paper is mainly tutorial, to describe mathematically and intuitively the fundamental relationships necessary to understand digital array processing. It is hoped that those readers with a background in antenna theory or array processing will see some of the advantages digital processing can offer, while those with a background in digital signal processing recognize the array processing area as a potential application for multi-dimensional signal processing theory.

157 citations


Journal ArticleDOI
J.A. Moorer1
01 Aug 1977
TL;DR: The use of analyis of natural sounds for synthesis, the use of speech and vocoder techniques, methods of artificial reverberation, theUse of discrete summation formulae for highly efficient synthesis, and the role of special-purpose hardware in digital music synthesis are discussed.
Abstract: The application of modern signal processing techniques to the production and processing of musical sound gives the composer and musician a level of freedom and precision of control never before obtainable. This paper surveys the use of analyis of natural sounds for synthesis, the use of speech and vocoder techniques, methods of artificial reverberation, the use of discrete summation formulae for highly efficient synthesis, the concept of the all-digital recording studio, and discusses the role of special-purpose hardware in digital music synthesis, illustrated with two unique digital music synthesizers.

150 citations


Patent
15 Dec 1977
TL;DR: In this paper, the error-correcting bits are added to the original bits in intersecting sets in row-by-column relationship, and parity bits can also be formed simultaneously with the formation of the error correcting bits as extensions of the intersectioning sets.
Abstract: Digital signals consisting of sets of simultaneous bits have an error-correcting signal encoded into them by adding an error-correcting bit to each set. The sets thus enlarged are referred to as digital words. The digital signals are then converted from simultaneous, or parallel, form to serial, or sequential, form and the digital words from a block of several digital signals at a time are interleaved in such a way that corresponding words from each of the digital signals in the same block are placed in immediate sequence. Prior to adding the error-correcting bits error-detecting bits can be added to the original bits in intersecting sets can that intersect the first-mentioned sets in row by column relationship, and parity bits can also be formed simultaneously with the formation of the error-correcting bits as extensions of the intersecting sets. In decoding the resulting signals, changes in the bits forming one word of each digital signal can be directly corrected, and additional errors can be detected and minimized by forming mean value signals of digital signals that immediately precede and follow the erroneous digital signal or by retaining the preceding digital signal until the succeeding signals return to correct or correctable form.

54 citations


Journal ArticleDOI
TL;DR: It is shown that a functional high-level language signal processing program can easily be modified so as to produce a similar program which, when executed, automatically generates another program containing precomputed algorithm sequencing and data access information.
Abstract: Optimal use of high-speed programmable digital signal processors generally demands familiarity with machine architectural features and hence, production of programs whose structure reflects and exploits those features. In contrast, it is apparent that little effort has been made to develop programming techniques which fully realize the signal processing computational capability of standard minicomputers. In this paper, it is shown that a functional high-level language signal processing program can easily be modified so as to produce a similar program which, when executed, automatically generates another program containing precomputed algorithm sequencing and data access information. The generated program will then utilize central processor arithmetic and logical capability only for data-dependent computation. In this way, instructions normally associated with computation for program sequencing/control or data access are eliminated, and all benefits of increased algorithm complexity for reduction of data-dependent arithmetic computation are in fact realized as decreased program execution time. Examples are given of Fortran programs which generate Fortran FFT subroutines and, for completeness, assembly language realizations of the Pfeifer/Blankinship autocorrelation algorithm. Results demonstrate that, using this technique, standard minicomputers may execute digital signal processing algorithms faster than peripheral processors which normally require standard minicomputers as host processors.

35 citations


Patent
18 May 1977
TL;DR: In this article, a digital signal processing circuit for controlling the spark timing and exhaust gas recirculation (EGR) of an internal combustion engine is presented, where the analog output signal proportional to the period of engine revolution is sampled by the same circuit and the digital output signal which is a function of the inverse of this analog input signal is effectively multiplied by the held pressure analog signal.
Abstract: A digital signal processing circuit for controlling the spark timing and exhaust gas recirculation (EGR) of an internal combustion engine is disclosed herein. The circuit first samples the magnitude of the vacuum manifold pressure and produces a complex function analog output signal in response thereto. The output signal is produced by first converting the vacuum pressure to a digital signal, then using a read only memory to produce a complex function digital output signal and then reconverting the digital output signal to an analog output signal. The pressure analog output signal is then stored in a first holding device. Subsequently, an analog signal proportional to the period of engine revolution is sampled by the same circuit. However, the circuit now develops a digital output signal which is a function of the inverse of this analog input signal and this inverse digital output signal is effectively multiplied by the held pressure analog signal to produce an EGR analog control voltage which is stored in a second holding device. Subsequently, an analog output signal which is a different inverse function of the engine period is developed and stored in the first holding device and then the EGR analog output signal is processed by the circuit and effectively multiplied by the analog signal now being held in the first holding device to produce a spark timing control analog output signal which is stored in a third holding device. The EGR control signal in the second holding device controls how much exhaust gas will be reinjected into the cylinders of the internal combustion engine and the spark timing control signal in the second holding device adjusts the timing of the sparks generated for the cylinders of the internal combustion engine.

25 citations


Journal ArticleDOI

24 citations


Journal ArticleDOI
TL;DR: In this article, a direct approximation technique of log magnitude response for digital filters is presented, which can be used to obtain the best mean-square approximation to an arbitrarily prescribed log-means response.
Abstract: A new direct approximation technique of log magnitude response for digital filters is presented in this paper. The facts that the log magnitude response of digital filters can be expanded into Fourier series and a fairly accurate cosine type log magnitude response can be realized by the elemental digital filter presented in this paper are used in the present technique. The system functions obtained by this method provide the best mean-square approximation to an arbitrarily prescribed log magnitude response. The resulting digital filters are realized in the cascade form of the elemental digital filters, and they give relatively low coefficient sensitivity. The elemental filter is recursive but its form is very simple. Its coefficients are easily obtained by the cepstrum of the impulse response which is the Fourier transform of the desired log magnitude response. This method is very powerful in the realization of digital filters for speech synthesis filters with complicated log magnitude responses.

18 citations


Patent
24 Jan 1977
TL;DR: In this paper, a system for tracking a pair of analog output signals from a resolver wherein such signals are hard-limited to provide square wave representations thereof, such hard limited signals are being supplied to at least one digital phase-locked loop in which the phases thereof are compared with those of the pair of square wave digitized feedback signals to produce an error signal.
Abstract: A system for tracking a pair of analog output signals from a resolver wherein such signals are hard-limited to provide square wave representations thereof, such hard limited signals being supplied to at least one digital phase-locked loop in which the phases thereof are compared with those of a pair of square wave digitized feedback signals to produce an error signal. The error signal is digitally integrated to produce a control signal for controllably changing the number of pulses in a pulsed clock signal which is supplied to a feedback counter for producing the pair of feedback signals having phases which are thereby changing so as to minimize the error signal.

18 citations


Proceedings ArticleDOI
G. Jullien1, W. Miller1, J. Soltis1, A. Baraniecka1, B. Tseng1 
09 May 1977
TL;DR: This paper discusses the application of the residue number system to realizing digital signal processing elements using such arrays and advantages and disadvantages over conventional realizations are discussed.
Abstract: In the past, hardware realization of digital signal processing elements have been based upon binary arithmetic concepts. Because of the dependence between digits in binary arithmetic operations, the hardware required to construct arithmetic elements is cumbersome. In the residue number system, arithmetic operations can be performed with complete independence between digits and a corresponding reduction in hardware complexity. In fact, using current technology, arithmetic operations can be carried out using arrays of look-up tables placed in high density ROMs. This paper discusses the application of the residue number system to realizing digital signal processing elements using such arrays and advantages and disadvantages over conventional realizations are discussed. Examples are given of recursive filter and FFT butterfly element realization.

Journal ArticleDOI
TL;DR: In this article, the use of near real-time digital signal processing to obtain acoustic data on the scattered form function of an object is considered and experimentally obtained backscattered form function for an infinite aluminum cylinder is presented for the region 4?ka?21.
Abstract: The use of near‐real‐time digital signal processing to obtain acoustic data on the scattered form function of an object is considered. Modifications inherent in the analog to digital conversion process are found not to alter the analog data representation. The experimentally obtained backscattered form function for an ’’effective’’ infinite aluminum cylinder is presented for the region 4?ka?21.

Patent
27 May 1977
TL;DR: In this paper, the phase of the transmitted signal is compared to a phase of a stable local oscillator and the phase relationship is converted to a complex digital number which is then stored in a suitable short term memory.
Abstract: An improvement in a pulsed radar system providing coherent-on-receive signals utilizing digital techniques. The phase of the transmitted signal is compared to the phase of a stable local oscillator and the phase relationship is converted to a complex digital number which is then stored in a suitable short term memory. The received signals are also converted to complex digital numbers containing both the phase relationship to the stable local oscillator and the amplitude of the return video signal. The digitized received signals are then digitally phase corrected to provide coherence using the stored complex digital numbers for the correction.

PatentDOI
TL;DR: In this article, the first and second sets of signals are combined in a piezoelectric crystal, and each signal in the first set is correlated only with its pair signals in the second set.
Abstract: An acousto-optic apparatus utilizing multiplexing techniques, for obtaininghe correlation of N pairs of signals. First and second sets of signals are combined in a piezoelectric crystal, and each signal in the first set is correlated only with its pair signal in the second set, that is, with the signal in the second set which has the same R.F. frequency. Each signal is a composite signal comprised of an envelope modulating an R.F. signal, and when a laser beam is directed across the interaction area of the crystal and directed onto a square law detector, the correlation of the envelopes of the N pairs of signals is obtained.

Patent
03 Feb 1977
TL;DR: In this article, a random access memory (RAM) accessed by counters and used for storing and shifting signals in a convolver, correlator, matched filter or multiplier is presented.
Abstract: A system for digital signal processing, including a random access memory (RAM) accessed by counters and used for storing and shifting signals in a convolver, correlator, matched filter or multiplier. An input signal and a reference signal are applied from opposite ends of the device such that the signals scan past each other at a relative velocity with respect to each other for obtaining either the convolution or correlation function of signals. The RAM operates as a shift register delay line and provides time scale inversion of signals when desired.

Journal ArticleDOI
M. Tasto1
TL;DR: The ‘PEAC’ structure and its application to various image-processing methods such as point operations, neighbourhood operations, guided boundary detection using prior knowledge, and object reconstruction from projections are discussed.
Abstract: A major drawback of digital computer image processing is the large computation time required. On the other hand, its flexibility, programmability and computational accuracy make it desirable to use digital processing. Advances in technology of LSI circuitry have now made it possible to increase strongly the computational power of image processing systems by combining many ‘micro computers’ or processing elements to array processors. We discuss several concepts of connecting such small computers and integrating them into a system, and then concentrate on the ‘PEAC’ structure which was closely investigated at PFH. Its application to various image-processing methods such as point operations, neighbourhood operations, guided boundary detection using prior knowledge, and object reconstruction from projections are discussed. In almost all applications a speed-up ratio of k can be achieved, where k is the number of processing elements.

Patent
25 May 1977
TL;DR: In this paper, the Fourier transform of multiple time-varying signals through electro-optical photoelastic means, photoconductive means, or photoemissive means was investigated.
Abstract: Signal processing apparatus having the capability to perform simultaneous space-time processing of sonar, radar and similar time-varying signals, and to effect the Fourier transform of multiple time-varying signals through electro-optical photoelastic means, photoconductive means, or photoemissive means.

Journal ArticleDOI
C. Baugh1
TL;DR: The design of the receiver exploits the use of subsampling techniques to increase the efficiency of the hardware through greater multiplexing and demonstrates the robustness of the digital signal processing techniques employed.
Abstract: An experimental multifrequency receiver for recognition of digitally encoded multifrequency signaling was designed, constructed and tested. The receiver is based on a quadrature detection technique that consists of digital demodulation followed by second-order, lowpass digital filtering. The post filtering processing produces an estimate of the amplitude of each of the six multifrequency tones and provides suitable information for thresholding and timing measurements. The receiver performs correctly even when subjected to severe environmental conditions including an analog signal range of 23 dB, 10 ms signal interruptions ('hits') and 20 ms signal spacings. The receiver's operation demonstrates the robustness of the digital signal processing techniques employed. The design of the receiver exploits the use of subsampling techniques to increase the efficiency of the hardware through greater multiplexing. When using subsampling, 128 multifrequency receivers with 16-bit words are realized with 6.5 dual-in-line packages per receiver; commercial TTL logic circuits, a 4-bit serial-parallel pipeline multiplier circuit, serial data and a 16.384 MHz clock are assumed.

Journal ArticleDOI
TL;DR: The procedure presented here is a transform domain approach that is distinct, to the knowledge of the author, when compared to known identification techniques in which a best fitting is made to an assumed mathematical model of the system.
Abstract: Algorithms for system identification and the computation of its mathematical model through a ``fast'' Z transformation of its sampled response in the presence of noise are introduced. It is shown that by iteratively applying constant-damping?and constant-frequency contour finite Z transforms a system's mathematical model?in the presence of noise can be efficiently evaluated. On line tracking of the poles and zeros of relatively rapidly time-variant systems such as a space shuttle or a jet aircraft are possible applications. An organization for a high-speed machine including a fast Fourier transform processor for on line identification of relatively rapidly time-variant system is suggested. Applications of the described algorithms include enhancement of poles in spectral analysis of signals, representation of signals by poles and zeros for signal classification, coding and recognition, filter synthesis, adaptive filtering, identification of parameters in curve fitting problems, in addition to system identification in the presence of noise. The procedure presented here is a transform domain approach that is distinct, to the knowledge of the author, when compared to known identification techniques in which a best fitting is made to an assumed mathematical model of the system. In addition to the smoothing obtained here through the computation of spectra in the Z plane of a time series including redundancy, no priori knowledge of the order of the system needs be assumed.

Patent
04 Nov 1977
TL;DR: In this article, an improved exposure time control for a photographic printer includes a digital processor such as a microprocessor, which receives input signals such as signals from large area transmission density (LATD) sensors, from density or color sensors, and from an operator control panel.
Abstract: An improved exposure time control for a photographic printer includes a digital processor such as a microprocessor. The digital processor receives input signals such as signals from large area transmission density (LATD) sensors, from density or color sensors, and from an operator control panel. Based upon the input signals, the digital processor derives a digital count and a clock control signal for each color channel. The exposure of each color channel is controlled as a function of the time required to change the corresponding digital count from its initial value to a predetermined final value. The changing of the digital count for each channel is caused by the digital processor in response to clock or interrupt signals from variable clocks controlled by the clock control signals. The rates at which the interrupt signals are generated are controlled by the clock control signals.


Patent
25 Oct 1977
TL;DR: In this paper, the authors presented an improved data processor architecture that combines the combination of an integrated circuit analog read-only memory, an analog alterable memory, and digital processing logic to achieve a low cost monolithic hybrid data processor.
Abstract: The present invention is directed to an improved data processor architecture. In a preferred embodiment, this architecture provides a hybrid stored program computer having analog and digital signals, where this architecture utilizes the combination of an integrated circuit analog read only memory, an integrated circuit analog alterable memory, and digital processing logic to achieve a low cost monolithic hybrid data processor. Analog CCD memories are provided for obtaining high capacity storage at low cost. Digital processing is provided for flexibility and capability. Analog to digital converters and digital to analog converters are used for communication between analog and digital portions of the data processor. Adaptive compensation having stored reference signals enhances analog signal precision.

Journal ArticleDOI
TL;DR: A digital way to produce an integral multiple of the frequency of periodic trigger pulses is described, used to generate the sampling pulses for the ADC of a time-series analyzer.
Abstract: A digital way to produce an integral multiple of the frequency of periodic trigger pulses is described. This procedure is used to generate the sampling pulses for the ADC of a time-series analyzer. The device which implements this method is called sampling frequency generator (SFG). A comparison is made between this digital method and the analog frequency multiplication method by means of a PLL.

Patent
07 Oct 1977
TL;DR: In this paper, the accuracy of the conversion of an analog signal to digital form is improved by amplifying the input signal by different factors and automatically utilizing the highest amplified input signal which has a voltage level within operational limitations in the analog to digital conversion process.
Abstract: The accuracy of the conversion of an analog signal to digital form is improved by amplifying the input signal by different factors and automatically utilizing the highest amplified input signal which has a voltage level within operational limitations in the analog to digital conversion process. Output data from a digital computer, based on the input data, is converted from digital form to analog form and is routed to a required location by the use of control logic and optically isolated digital signals used to address other computers.

Proceedings ArticleDOI
28 Dec 1977
TL;DR: This paper describes the development of a new CODP input device capable of the required speed and spatial bandwidth and employs an integral CCD input to accept serial electronic signals and a liquid-crystal light valve output for C ODP.
Abstract: In the face of the increasing need to be able to process two-dimensionally organized data, the present electronic digital processing and coherent optical data processing techniques have proven inadequate. The potential advantage of coherent optical data processing (CODP) is that, as opposed to the serially organized electronic digital computers, it can simultaneously process data on a very large number of parallel channels. The limitation to the usefulness of CODP has been the lack of a high-speed, high-spatial-bandwidth input device. This paper describes the development of a new CODP input device capable of the required speed and spatial bandwidth. The device employs an integral CCD input to accept serial electronic signals and a liquid-crystal light valve output for CODP.© (1977) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Patent
07 Apr 1977
TL;DR: In this article, the authors proposed to make it possible to generate a received beam with random characteristics easily and economically with high precision by applying digital signal processing technique, where the data are sequentially read out at a rate equal to the sampling rate of the input side.
Abstract: PURPOSE:To make it possible to generate a received beam with random characteristics easily and economically with high precision by applying digital signal processing technique. CONSTITUTION:Input signals of K channels are sampled at a constant rate and converted by A-D converters 221-22k into digital codes, which are written in input memories 231-23k. On the other hand, data block consisting of N sampled values read from memories 231-23k are sent to FFTs processors 241-24k, where the next Fourier transformation comes into effect to calculate complex spectrums as many as N/2. Next, transfer functions in table memories 261-26k are multiplied by complex multipliers 251-25k by the above-mentioned spectrums as to every frequency component and the results are added by complex adder 27 covering all channels; and the results are Fourier-transformed reversely by FFT processor 28 and the result is written in output memory 29. The data are sequentially read out at a rate equal to the sampling rate of the input side.

Proceedings ArticleDOI
01 May 1977
TL;DR: A novel concept for digital signal processing using universal hardware and flexible software is presented, which offers multiprocessing facilities with multiple function blocks for data and structured data processing.
Abstract: A novel concept for digital signal processing using universal hardware and flexible software is presented. A universal signal processor (USP) offers multiprocessing facilities with multiple function blocks for data and structured data processing. The basic components such as the CROSSBUS, the input/output processor, the data and structured data processor and the instruction processor are build with standard microprocessor slices. The concept of software is based on the high level language SIPROL (signal processing language) which is derived from PASCAL. The main differences with respect to the manifold problems in digital signal processing are outlined. Examples are presented which illustrate the proposed concept.

Proceedings ArticleDOI
08 Dec 1977
TL;DR: The approach taken in this study was to determine a computationally efficient algorithm, verify its theoretical performance relative to the conventional multiply and sum correlation procedure, and to estimate the hardware resources necessary to compute the recommended algorithm in real-time.
Abstract: A study has been conducted to show the feasibility of implementing an all-digital correlator for missile terminal area guidance. The central thrust of this effort was to establish the hardware requirements for realizing multiple area cross-correlations in real-time using modern digital signal processing techniques. The ultimate objective of the study is the improvement in terminal accuracy of long-range Army missiles through digital area correlation guidance. The approach taken in this study was to determine a computationally efficient algorithm, verify its theoretical performance relative to the conventional multiply and sum correlation procedure, and to estimate the hardware resources necessary to compute the recommended algorithm in real-time. The algorithm recommended for implementation is the high speed digital correlation algorithm which uses the fast Fourier transform (FFT) to minimize the total number of arithmetic computations. The computational equivalence of the high speed correlation algorithm to the conventional multiply and sum approach was demonstrated by example using a digital computer program and simulated two-dimensional test data. A specific all-digital correlator hardware design has been postulated and documented at the block diagram level. This design was used to estimate the number of integrated circuits, as well as the power and space requirements, of an all-digital area correlator.© (1977) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Patent
16 Dec 1977
TL;DR: In this article, the analog signals are compared with reference levels based on the average level of the analog signal, for instance, alternating components of analog signals compared with a zero level, thereby improving accuracy of the conversion by minimizing the effect of DC and gain drift.
Abstract: In an analog to digital converting device having plural converting stages provided for converting an analog signal to a multi-bit digital signal, the lowest or just preceding converting stages, wherein analog signals to be converted are quantized by minute steps and can be regarded as similar to Gaussian noises, are arranged to form equivalent digital signals. The analog signals are compared with reference levels based on the average level of the analog signals, for instance, alternating components of analog signals are compared with a zero level, hereby improving accuracy of the conversion by minimizing the effect of DC and gain drift.

Patent
25 May 1977
TL;DR: In this paper, the receiver recognises and evaluates analogue signals having a specified frequency and is for use in communication and data processing systems, and converts the received analogue signals into digital signals.
Abstract: The receiver recognises and evaluates analogue signals having a specified frequency and is for use in communication and data processing systems. A device samples the received analogue signals and converts them into digital signals. A microprocessor is connected to this device and processes the digital signals by the quadrature correlation method controlled by a program. The specified frequency signals are detected. These detected signals are evaluated by a further device or by the microprocessor. Either single sinusoidal signals of specified frequency or multifrequency coded signals may be evaluated.