Showing papers on "Digital signal processing published in 1972"

Effects of finite register length in digital filtering and the fast Fourier transform

Abstract: When digital signal processing operations are implemented on a computer or with special-purpose hardware, errors and constraints due to finite word length are unavoidable. The main categories of finite register length effects are errors due to A/D conversion, errors due to roundoffs in the arithmetic, constraints on signal levels imposed by the need to prevent overflow, and quantization of system coefficients. The effects of finite register length on implementations of linear recursive difference equation digital filters, and the fast Fourier transform (FFT), are discussed in some detail. For these algorithms, the differing quantization effects of fixed point, floating point, and block floating point arithmetic are examined and compared. The paper is intended primarily as a tutorial review of a subject which has received considerable attention over the past few years. The groundwork is set through a discussion of the relationship between the binary representation of numbers and truncation or rounding, and a formulation of a statistical model for arithmetic roundoff. The analyses presented here are intended to illustrate techniques of working with particular models. Results of previous work are discussed and summarized when appropriate. Some examples are presented to indicate how the results developed for simple digital filters and the FFT can be applied to the analysis of more complicated systems which use these algorithms as building blocks.

Discrete representation of signals

Abstract: In processing continuous-time signals by digitalmeans, it is necessary to represent the signal by a digital sequence. There are many ways other than periodic sampling for obtaining such a sequence. The requirements for such representations and some examples are discussed within the framework of simulating linear time-invariant systems. The representation of digital sequences by other digital sequences is also discussed, with particular emphasis on the use of such representations to implement a nonlinear warping of the digital frequency axis. Some applications and hardware implementation of this digital-frequency warping are described.

A Simple Approach to Digital Signal Processing

Abstract: From the Publisher: Digital signal processing (DSP) has been a very active area of research and application for over thirty years Applications range from audio and video electronics to sensors, image processing and medical applications Provides a basic introduction and simple overview of DSP

Teminology in digital signal processing

Abstract: The committee on Digital Signal Processing of the IEEE Group on Audio and Electroacoustics has undertaken the project of recommending terminology for use in papers and texts on digital signal processing. The reasons for this project are twofold. First, the meanings of many terms that are commonly used differ from one author to another. Second, there are many terms that one would like to have defined for which no standard term currently exists. It is the purpose of this paper to propose terminology which we feel is self-consistent, and which is in reasonably good agreement with current practices. An alphabetic index of terms is included at the end of the paper.

Monopulse radar receiver with error correction

Abstract: A monopulse radar receiver is disclosed wherein angle error digital computation apparatus is used for correcting received target return signals in accordance with digital correction signals which are derived within such receiver in response to pilot pulses. Such digital computation apparatus includes means for generating, from pilot pulses, digital signals representative of the pair of quadrature components of video signals developed in the sum channel and each one of the difference channels. The quadrature components associated with the target return signals are then corrected in accordance with the quadrature components associated with the digital correction signals.

Digital diversity combiner

Abstract: This relates to a predetection maximal ratio digital diversity combiner for a phase shift keyed digital data signal propagating on N different paths through a dispersive medium, where N is an integer greater than one. Each of N signal channels respond to the data signal propagated on a different one of the N different paths. Each of the channels include an arrangement to separate the data signal into an inphase component and a quadrature component and also a pair of analog-to-digital converters to convert the inphase component into an inphase digital signal and the quadrature component into a quadrature digital signal. A digital adder arrangement is coupled in common to the output of each of the N channels to digitally add the inphase digital signal of each of the channels together to produce a combined inphase digital signal and to digitally add the quadrature digital signals of each of the channels together to produce a combined quadrature digital signal. A decision circuit responds to the most significant digit of both the combined inphase digital signal and the combined quadrature digital signal to recover the data conveyed by the data signal. A clock recovery circuit responds to the combined inphase digital signal, the combined quadrature digital signal and the recovered data to produce properly phased timing signals for control of the decision logic, each of the analog-to-digital converters and an automatic gain control circuit common to each of the N channels. Each of the channels further include an arrangement coupled between the associated pair of analog-to-digital converters and the digital adder arrangement and also to the decision circuit. This arrangement is responsive to the recovered data and the inphase and quadrature digital signals to determine the maximal ratio weights of these signals. The determined inphase and quadrature digital weight signals are employed to weight the inphase digital signal and the quadrature digital signal prior to digitally adding thereof in the adder arrangement. An automatic gain control circuit is coupled to the last mentioned arrangement of each of the channels and to the clock recovery circuit to produce an automatic gain control signal to control the gain of the data signal in each of the channels. This is accomplished by detecting the maximum maximal ratio weight of either the inphase or quadrature digital signal of any of the channels involved in the diversity combiner and generating from this maximum maximal ratio weight an automatic gain control voltage.

Signal converting systems for use in stereo reproducing systems

Abstract: A signal converter is provided for intermixing first and second audio signals reproduced from a conventional 2-channel recording medium to produce two difference signals which are supplied to a 2 to 4-channel converter for improving the separation between the front and rear signals. The signal converter may be constructed to mix together first and second audio signals at a variable relative amplitude ratio therebetween and with a variable polarity relationship to form two sum signals or two difference signals.

Synthesis of digital signals corresponding to selected analog signals

Abstract: Digital signals for use in a time-divided multiplex signalling system are synthesized directly in digital form. Binary signals indicative of the differences between the values of successive time-spaced samples of preselected analog signals are fed through gates to an up-and-down accumulator under control of a counter and phase and polarity discriminators. The output of the accumulator at any instant represents the algebraic sum of all previous signals received by it. The preselected analog signals are preferably of the kind that can be represented by the sums of one or more simple trigonometric functions, so the sample values need be calculated only for a quarter wave.

Signal mode converter and processor

Abstract: An analog signal comparator is coupled to provide its output to a data processor which has digital outputs coupled through a digital-to-analog converter to an input of the comparator, for periodically producing in the processor digital representations of samples of another analog signal that is also applied to the comparator. In each sampling period, additional data processing is carried out with respect to the digital representations; and an output circuit coupled to the converter output employs a filter circuit to select only the results of the last-mentioned processing. The additional processing is shown for digital filters featuring single and tandem sections.

Electronic musical instrument with digital reverberation system

Abstract: An electronic musical instrument with a digital reverberation system in which one of the digital signals in a pair of digital signal information channels is delayed. The system is particularly adapted for use in an electronic musical instrument in which a digital multiplex signal is generated by actuation of the keys of the instrument. The digital multiplex signal is used to address a digital wave shape memory which stores signals representative of a musical note in digital form. The signals read out of the digital wave shape memory are, after processing, converted to an analog signal by means of one or more digital to analog converters. A pair of digital channels, one of which has a digital delay means therein, may be incorporated into an electronic musical instrument at various stages. The delay may be incorporated into one of the digital channels before or after the reading of the digital wave shape from memory. In addition, means may be provided to selectively switch between the delayed and undelayed signals as desired for the pedals, great or lower manual, and the swell or upper manual by means of a multiplex keyboard selector which in conjunction with an electronic switch selectively switches in the delay for the portion of the multiplex signal corresponding to a particular keyboard.

A survey of digital speech processing techniques

Abstract: Digital signal processing techniques are becoming increasingly important in speech analysis and synthesis. These techniques can be implemented using a general purpose computer facility (often not in real time), or special purpose hardware realizations can be constructed. This paper discusses some recent work in speech processing including design of digital filter bank spectrum analyzers, homorphic analyzers of speech, predictive coding, and hardware realization of a digital formant synthesizer. The survey concentrates on those speech processing techniques relevant to the development of sensory aids for the deaf.

Synchronizing and multiplexing in a digital communications network

Abstract: A concept of a totally digital communications network is described in which all signals are converted into digital form and remain so as they are multiplexed, switched, and transmitted. To permit time-division techniques for such network processing, and thus realize significant economic advantages, the digital signals must be either generated in synchronism or brought into synchronism. While the switching portion of this network is not yet in service, some of the transmission network already exists and the evolution toward a digital transmission hierarchy is well underway. The hierarchy consists of 1) terminals which convert analog Signals into digital form suitable for transmission, 2) transmission facilities which are available at various capacities, and 3) multiplexers which can derive several lower capacity digital facilities from a single high-capacity system. Such a network, when complete, can also disseminate time and frequency for other uses with the same accuracy as required by the network. The network requires that the relative phase difference between any two signals must be bounded, which means exact matching of the long-term average frequency throughout the network. Signal-processing time and variations in propagation velocity in various media-cable, radio, waveguide-control the short-term accuracy. This network can thus provide time-and-frequency information proportional to the observation time. Available techniques that can achieve such accuracy for the network are discussed.

Signal detectability performance of optimum Fourier receivers

Abstract: FFT processing systems have become important and extremely useful implementations of narrow-band signal detectors for signals with unknown phase and frequency This paper examines the Fourier filter and detector and shows it to be an optimum receiver for a very generalized signal model The general signal model covers a single sinusoidal pulse, pulse trains, and continuous signals The model provides for the finite bandwidth encountered in real signals and includes the affect of redundant processing which can be employed with high-speed FFT processors The paper provides a method for evaluating detection sensitivity for this class of signal detectors and gives experimental results from an FFT processor implementation

Television multiplexing system

Abstract: A television multiplexing system which includes a circuit that inserts a digital coded sync signal and a digital code into a video signal for identifying the channel from which the video signal was generated so that a plurality of signals can be sent over a single hard-line The digital sync signal and the digital coded signals are generated by a single crystal controlled clock so that they are always in synchronism with each other In demultiplexing the signals so as to feed the video signal to a proper recording channel the sync signals are utilized for shifting the digital coded signals into a shift register and the shift register, in turn, activates a decoder according to the code stored in the shift register for selecting the proper recording disc or receiver for storing the video signal

Passive image stabilization system

Abstract: An image tracking sensor system mounted to an inertial platform generates an analog image displacement signal for passive image stabilization of a scene image viewed from the platform. A radiometric signal having a frequency proportional to the relative motion between the scene image and a reticle is generated by the convolution of the image with the reticle. This signal is detected and electronically processed in a DC squareroot amplifier, a threshold decision logic and an analog to digital converter which generates a digital signal related to image velocity. The digital signal is measured by a digital differential analyzer doppler frequency comparator which generates digital signals defining image velocities relative to boresight. These digital velocities are integrated and displacements accumulated in a digital up-down counter tracking register. The accumulated displacements represent changes of image position relative to boresight. The accumulated digital displacements are read out of the tracking register by means of a digital to analog converter. An analog image displacement signal generated by the digital to analog converter is applied to the inertial platform for gyro-torquing stabilization control thereof.

Phase representative digital signal modulating apparatus

Abstract: Apparatus for combining two series of digital numbers from each of two different sources wherein the numbers represent phases of signals and for providing an output indicative of the modulation of one of the signals by the other signal with the output being an analog or digital number signal. The concept generally is shown in detail as applied to a differential phase shift keying transmitter where one of the sources of digital numbers is a time division multiplexed source of two signals and the other is a time base or clocking signal which modulates each of the time multiplexed signals from the first source on a raised cosine basis through the use of a look-up table in the form of a read only memory.

Circuit arrangement for analog-to-digital conversion of magnitudes or signals in electrical form

Abstract: Apparatus for converting analog electrical signals to a digital electrical signal wherein the statistical probability of a predetermined state for the digital output signal is a predetermined function of the analog input signals to provide digital signals having improved transmission capabilities. A continuously varying threshold electrical signal having a predetermined statistical distribution of amplitudes is compared to the analog input signal and the result of such a comparison is used to produce and control the digital output signal such that the statistical probability of a given digital state is a predetermined function of the analog input corresponding to the predetermined statistical distribution of amplitudes of the continuously varying threshold signal.

Multirate digital filtering

Abstract: The concept of multirate digital filtering is introduced here. Two specific schemes are under scrutiny, one consisting of a digital prefilter and the other consisting of an analog prefilter.

Recognition system and processor

Abstract: A recognition system for viewing and processing information on an article such as a letter or the like including an optical system for viewing the information, a diode array for converting the viewed image of the information into analog electrical signals, an amplifier for amplifying the analog electrical signals, an input conditioner for converting the amplified analog electrical signals into digital signals and a processor for processing the digital signals to compensate for errors which might be present in the information on the article and in the analog and digital signals.

Digital filtering for radar signal processing applications

Abstract: A novel approach in synthesizing digital filters for signal processing applications is presented. This approach is an extension of the frequency sampling method of nonrecursive filter synthesis. Appropriate time delays are used in conjunction with a set of parallel complex exponential resonators whose outputs are summed to yield a desired filter impulse response. This synthesis method takes advantage of the known signal waveform structure and results in many fewer digital computations as compared to convolution processing. This approach is particularly suited to synthesis of matched filters for radar signal processing and yields matched or approximately matched filters which simultaneously have very low storage and very low computational requirements.

A Theory for Optimal MTI Digital Signal Processing: Part 1. Receiver Synthesis

Abstract: : A classical problem in radar theory is the detection of moving targets in a ground clutter plus receiver noise background. Improvements in clutter rejection have recently been made by replacing analog MTI processors by their digital equivalents as this eliminates many of the problems associated with the maintenance of the analog hardware. In an attempt to determine the ultimate improvements possible using this new technology, the MTI problem was formulated as a classical detection problem and solved using the generalize likelihood ratio test. By manipulating the likelihood ratio, the receiver could be interpreted as a clutter filter in cascade with a doppler filter bank. The performance of the optimum receiver was evaluated in terms of the output signal- to-interference ratio and compared with well-known MTI processors. It was shown that near-optimum performance can be obtained using a sliding weighted Discrete Fourier Transform (DFT).

Randomly sampled digital filters

Abstract: The concept of a randomly sampled digital filter is introduced and formulated in this paper. An equivalent problem, which is of utmost importance in digital filtering, is also considered, namely, the problem of digital filtering with faulty samplers. Error analysis is carried out in two different approaches: 1) frequency response method, and 2) worst-case analysis. With the confirmation of experimental verification, very simple formulas are derived that are applicable under different realistic assumptions, and are most appealing to practical design purposes.

Zoom FFT - An Approximate Vernier Frequency Algorithm.

Abstract: : This report describes a method for obtaining fine frequency resolution. The technique employs the partitioning of a large data sequence by performing small size FFTs and other digital signal processing techniques to achieve the finer resolution. The method yields approximate results. The accuracy appears to be dependent on Vernier bandwidth, signal-to-noise ratio and signal spectrum. A FORTRAN program is available in the Appendix. (Author)

Signal Processing Element Functional Description. Part 2 (Preliminary). Signal Processing Arithmetic Unit

Abstract: : The NRL Signal Processing Element (SPE) is being developed to provide a high-performance signal processing facility for radar, sonar, and communication systems It is intended to be compatible with the Navy's All Applications Digital Computer (AADC) The SPE consists of four major subsystems; a Microprogrammed Control Unit (MCU), a Buffer Store and Storage Control Unit (SPU), a Signal Processing Arithmetic Unit (SPAU), and Input/Output (I/O) units

Use of the geometry of partial sums in digital filter analysis

Abstract: The z transform of the truncated impulse response of a digital filter with poles is a polynomial that is important in the stability analysis of certain types of block and inverse digital filters. The geometry of partial sums is the theory that describes the behavior of the zeros of these polynomials. This paper presents specific results and also develops guidelines for use in stability analysis and design of block and inverse digital filters.

Digital processing of marine seismic data

Abstract: The enormous amount of marine seismic data makes it necessary to process the data with a high speed digital computer. This paper is concerned with the application of digital signal processing techniques to the marine seismic data. The objectives of the study are two-fold: (1) to improve the data, i.e. to enhance the signal-to-noise ratio of the data and (2) to reconstruct the seismic profile from the digitally filtered data for a better interpretation of the ocean bottom and the subbottom structure. Cepstrum analysis, deconvolution, and Walsh domain processing of the data are discussed and the unpublished computer results are presented in detail.

A Survey of Di Speech Processi Techniques

Abstract: Digital signal processing techniques are becoming increasingly important in speech analysis and synthesis. These techniques can be implemented using a general purpose computer facility (often not in real time), or special purpose hardware realizations can be constructed. This paper discusses some recent work in speech processing including design of digital filter bank spectrum analyzers, homorphic analyzers of speech, predictive coding, and hardware realization of a digital formant synthesizer. The survey concentrates on those speech processing techniques relevant to the development of sensory aids for the deaf.

A Signal-Dependent Error Arising in Digitally Processed Images Due to Quantization

Abstract: A troublesome error has been found in digital data which represents an image renormalized by a computer. The error, which takes the form of anomalous patterns in the switching sequence of certain bits, contributes unnecessary, signal-dependent noise to the-data. Experimental results illustrating this phenomenon are shown and a theoretical model is derived. The model shows that the phenomenon results from a qtiantization or roundoff error inherent in digital renormalization and describes its detailed behavior. A method for rescaling the data which does not lead to the introduction of noise is described.

The complex digital filter and its applications in digital signal processing

Abstract: Digital computer simulation of communication systems have been gaining wide acceptance and usage as a tool for analysis. In some cases, when the number of independent parameters is large or the processes are highly nonlinear, it is the only viable technique. In most digital computer simulations, the digital representation of bandpass filters can impose serious synthesis problems when conventional digital filter synthesis techniques are utilized. It is shown that the use of a complex (real and imaginary) technique of digital filter synthesis can eliminate several of the synthesis problems associated with conventional techniques. Three applications of the com lex technique are described in this paper.