Showing papers on "Digital signal processing published in 1974"

TDM-FDM Transmultiplexer: Digital Polyphase and FFT

Abstract: The cascading of a discrete Fourier transform processor and a digital polyphase network is shown to reduce the computation rate in frequency multiplexing-demultiplexing systems to a value close to minimum. Implementation advantages of that technique are pointed out. The highest efficiency is achieved when the number of channels is close to a power of 2, which is demonstrated by the 60-channel frequency-division-multiplexing (FDM)-2 × 30-channel time-division-multiplexing (TDM) transmultiplexer; a rough estimate of the computation rate is carried out in a practical case and appears to be quite within reach of the present technological capabilities. Significant cost advantage over equivalent analog equipment is expected. A digital version of the 12-channel FDM system using the same technique is also considered.

Video system for conveying digital and analog information

Abstract: A video system having a transmitter in which a composite signal is formed by combining digital information with scanlines of an analog video signal generated by a line scanning device, and a receiver in which the digital information is recovered. The digital information is combined with the analog video signal at predetermined locations along scanlines of the video signal, and these predetermined locations are varied in order to prevent visible deterioration of the video image. In the video receiver, the digital information is recovered by examining the composite signal at the predetermined locations to extract the digital information. Each bit of digital information to be conveyed is represented by a first pseudo-random digital pulse sequence (or its complement, depending on whether the data bit is 1 or 0) which is superimposed on a selected scanline of the analog video signal to form the composite signal. The digital information is recovered at the receiver by generating a second pseudo-random digital pulse sequence in synchronism with the first sequence, and by examining the composite signal at locations determined by the second digital pulse sequence to extract the digital information contained in the composite signal.

Fast one-dimensional digital convolution by multidimensional techniques

Abstract: This paper presents two formulations of multi-dimensional digital signals from one-dimensional digital signals so that multidimensional convolution will implement one-dimensional convolution of the original signals. This has reduced an important word length restriction when used with the Fermat number transform. The formulation is very general and includes block processing and sectioning as special cases and, when used with various fast algorithms for short length convolutions, results in improved multiplication efficiency.

Image Processing In Signal-Dependent Noise

Abstract: Image detection noise is a fundamental limitation in picture processing, whether analog or digital. This noise is characteristically signal-dependent and this signal-dependence introduces significant problems in the design of appropriate noise-suppression techniques. This paper outlines some recent results obtained by the authors in the optimum suppression of two types of signal-dependent image noise: film-grain noise and photoelectron shot noise. The work in grain noise suppression involves deriving the minimum-mean-square error Wiener filter for a new form of signal-dependent noise model suggested in earlier work by T. S. Huang. Implementation of these filters by either coherent optical or digital processing techniques is possible. Digital computer simulations of grain noise suppression using two particular cases of this additive, "signal-modulated" noise model were performed. They demonstrate the potential advantages of noise suppression filters which make use of a priori knowledge of the signal-dependent nature of the grain noise. The results of work on linear, unbiased restoration of images recorded in the presence of photoelectron noise are summarized. Additional work in both of these areas is suggested, with a particular need existing for correlating the properties of various models proposed for grain noise with experimental data obtained on emulsions using scanning microdensitome ters.

Apparatus and method for testing digital circuits

Abstract: A device is disclosed which converts a digital signal or bit stream into a digital signature repesentative of the digital signal by means of a feedback shift register. The apparatus may be used to identify and characterize digital signals at various test points in an apparatus for testing purposes. Signatures for digital signals from properly operating circuits can be recorded in a variety of fashions for later comparison with signatures of digital signals from circuits under test. The comparison of the signatures enables a person using the apparatus to determine whether the circuit under test is operating properly and, if it is not, to locate the fault in many instances. The apparatus may also be used to examine digital signals to enable identification of transient errors.

Digital storage system for high frequency signals

Abstract: A system and a method are provided for storing high frequency signals, for example microwave signals, for later reproduction with any desired degree of fidelity. The system and method involve determining a measure of the phase difference between an incoming high frequency signal and a converting signal. The measure of the phase difference consists of a plurality of signals which may change rapidly. The phase difference signals are converted to digital frequencies and may be stored or transmitted in that form. Later, the digital signals may be converted to analog signals (if necessary) and used to control the phase of a signal having the frequency of the converting signal. This process reconstructs the incoming high frequency signal.

Speech Interpolation in Digital Transmission Systems

Abstract: In a telephone conversation a channel is activated by the voice only 25 percent of the time. It is therefore possible to provide a number of m connections with a number of n channels, where n is less than m . When the number of connections is large, m/n (gain) tends to the inverse of the activity. The interpolation procedures used in overload conditions and the resulting degradations, as well as the typical circuits of a voice interpolation apparatus, are examined. Also, two very highly interesting developments are herein illustrated, the ATIC equipment and a single-channel digital speech interpolation (DSI) developed only for measurement purposes. The results obtained from measurement studies carried out with the aforementioned apparatus have enabled the determination of the criteria which must be followed in the selection of the most suitable interpolation procedures. Besides examining the complexity and the reliability which the DSI equipment calls for, it has been possible to analyze the precise working characteristics under normal operating conditions.

Signal correction system

Abstract: In a system producing stored digital words representing an optical image, a digital error correction system automatically corrects the video signal for shading. A television camera converts the optical image of a laboratory microscope slide into an analog electrical signal representing the optical image along a raster of scan lines. An analog-to-digital converter produces digital words representing the magnitude of the video signal at periodic times along each scan line. These words are stored to represent the optical characteristics of the image. In order to correct the analog video signal for shading error, digital words representing the pattern of the shading are stored in a digital memory. This memory is loaded during intervals in which the analog signal represents only the shading. During this time, the video signal is converted to digital words and stored in the digital memory. Thereafter, during normal operation, these digital words are converted to an analog error signal which is subtracted from the video signal.

A Microprogrammed Approach to Signal Processing

Abstract: This paper provides a summary of the design features of a microprogram controlled high speed signal processor referred to as an analyzer unit (AU). It provides an overview of the interrelationship between processing requirements, a language structure in which these requirements can be efficiently programmed, and an AU design that can efficiently execute the processing requirements. Particular emphasis is placed on the microprogram structure of the system computational unit, the Arithmetic Processor.

Signal editing and processing apparatus

Abstract: A signal editing and processing technique for converting a plurality of continuous signals, especially long term continuous audio signals respectively relating to short term still picture signals, to a transmitting signal in which signal transmission periods and pause periods are provided, having an integer ratio of time duration with each other, wherein other signals, especially the picture signals, should be transmitted. All continuous signals are sequentially converted to digital signals addressed in accordance with relevant continuous signals, and once stored in arbitrary positions of a memory, and then read out in a given multiplexed sequence corresponding to the transmission periods of the transmitting signal. The read out multiplexed digital signals are sequentially stored in another memory, and then read out with a given high speed equal to that of the signal transmission. The digital signals read out with the high speed are stored in still another memory, and then read out repeatedly to form the transmitting signal.

On Linear Receivers for Digital Transmission Systems

Abstract: Two main classes of receivers for data modems using linear modulation systems over time-dispersive channels have been investigated by many authors for both theoretical and practical purposes, 1) structure-constrained linear receivers, such as zeroforcing and mean-square-error tapped-delay-line equalizers, and 2) nonlinear receivers, such as decision-feedback equalizers and maximum likelihood sequence estimators. In this paper a linear receiver that turns out to be practical and optimum in the mean-square sense is analyzed in detail, and some interesting features of this receiver are stressed; for instance, it is shown that in the absence of noise it becomes a zero-forcing equalizer, provided that stability can be achieved. A comprehensive set of results is also presented, showing that conventional tapped-delay-line equalizers perform very close to the optimum.

Readback circuits for digital signal recorders

Abstract: Digital signal processing techniques enhance readback of digital signals from a magnetic recorder. Sampled signal amplitudes obtained at bit period boundaries and midpoints are converted to digital signal sets, which sets are processed for equalization, DC restoration, phase error detection, generation of timing signals, and amplitude compensation of the sampled signal amplitudes.

An Optical-Digital Image Processing System

Abstract: The combination of optical preprocessing and interactive digital processing furnishes a powerful technique for image analysis and pattern recognition. We describe in this paper several innovative data-reducing operations easily implemented with an optical system. A particular design for an interactive optical-digital computer now being assembled is discussed and several actual examples of optical preprocessing are given.

Digital matched filter and correlator using random access memory

Abstract: A system for digital signal processing, including a random access memory accessed by counters and used for storing and shifting signals in a time compressor matched filter or correlator

Updated diver navigation device

Abstract: In a diver's navigation system employing a receiver having three equal, alarly disposed hydrophone locations, a digital signal processing circuit is employed which receives frequency shift keyed digital data from remote beacons. Circuitry is provided to process this data to derive a bearing indication to the beacon.

Digital phase-locked loop generating signed phase values at zero crossings

Abstract: Digital signal processing techniques are employed to construct a digital phase-locked loop (PLL). The PLL of the present invention is advantageously employed in a readback circuit for a digital signal magnetic recorder. A unique phase error detector stores at least three digital values representative of three successive samples of a periodic waveform to which the PLL is to be synchronized. Upon detection in a change of sign of the digital values, a transition is indicated. One of the three numbers, preferably the center one, is transferred to a PLL digital filter wherein the successive values are averaged and scaled for controlling a digital VFO, also referred to as an NCO (numerically controlled oscillator). The digital VFO has a pair of counters which count down to zero, then emit a transition generating a clock signal for synchronously operating the sampler and other circuits in the readback apparatus.

Digital Companding Techniques

Abstract: This paper deals with the requirements for the design of digital companding techniques in either delta or pulse-code modulation. Both delta and pulse-code modulation convert analogue signals into binary signals and in both these systems the dynamic range is normally small. By the use of companding, the dynamic range can be extended. Since both delta and pulse-code modulation are digital methods, they are well suited to the use of digital companding techniques. The binary transmitted signal normally contains a measure of the system performance. By observing certain patterns in this binary signal and using the occurrence or nonoccurrence of these patterns to change the gain of the modulator and demodulator, syllabic companding can be obtained. The selection of the binary pattern and the rate of change of gain of the modulator and demodulator, determines both the point at which the companding operates and the attack and decay times. The ratio of the largest to the smallest value of the gain determines the dynamic range. By the use of digital circuitry, the gain can be controlled with sufficient accuracy over a large dynamic range. The paper deals with the principles involved in selecting the binary patterns to control the gain of the modulator and as examples a delta modulation system and a pulse-code modulation system with companding ratios of 60 dB are discussed.

Digital FFT processor using random access memory

Abstract: A system for digital signal processing including a random access memory accessed by counters and used for storing and shifting signals in a fast Fourier transformer (FFT).

Predictive conversion between self-correlated analog signal and corresponding digital signal according to digital companded delta modulation

Abstract: A predictive analog-to-digital converter for converting a selfcorrelated analog signal into a delta modulated digital signal according to companded delta modulation comprises a digital step size signal generator responsive to the digital signal for producing a digital step size signal variable to represent at least three step sizes for successive quantization of the selfcorrelated analog signal, a memory for memorizing a digital sum signal supplied thereto and for producing the memorized digital signal, an adder for deriving the algebraic sum of the memorized digital signal and the digital step size signal to produce the digital sum signal to be newly supplied to the memory, and a local digital-to-analog converter for converting the memorized digital signal into the predicted analog signal for use in comparison with the self-correlated analog signal in accordance with the predictive conversion technique.

Method and apparatus for mixing and recording multi-track stereo audio signals which have been recorded as several individual audio signals

Abstract: A method of and apparatus for mixing and recording multitrack stereo audio signals from individual audio signals while utilizing the development of analog signals which are indicative of level, etc., converting the signals to digital form for clock controlled processing and reconverting to analog form to actuate mixing control elements.

Dual tone multiple frequency receiver/decoder

Abstract: A method and apparatus for separating the two analog component signals which make up a dual tone multiple frequency (Touch Tone) signal, converting the component signals into digital identification signals representative of the component signals, and then recombining the identification signals and developing a digital output signal which corresponds to the particular input signal. The apparatus includes signal separating means for separating the input analog signal into its two component signals, frequency identification decoding means for identifying the two component signals and developing digital identification signals corresponding thereto, and digital logic means responsive to the identification signals and operative to develop particular one-of-sixteen digital output signals which correspond to the particular Touch Tone signal input to the system.

Circuit for curtailing effects of bit errors in pulse coded transmission

Abstract: A differentially pulse coded digital representation of a continuous analog signal is digitally accumulated, and each time polarity information indicates that the accumulated digital approximation changes sign the digital representation is complemented. The result of this action is that each bit of a predetermined type in the digital representation, and following the complementing point in a signal flow sense, has the same directional effect, on the digital accumulation, with respect to a predetermined signal reference level within the range of the continuous analog signal variation. Several embodiments are shown with different types of digital accumulation and different circuit locations for realizing the inversion of the digital representation.

Digital time measurement system

Abstract: This invention relates to the digital measurement of times and time intervals. An object is to provide time measurement data in a form suitable for input to a digital computing system. In a particular form of the invention, a number of oscillators or signal sources are synchronized so that their output frequencies are related exactly as successive powers of two, and their output signals are maintained in a predetermined phase relationship. For example, a set of oscillators may be arranged to generate signals of 1 MHz, 2 MHz, 4 MHz, 8 MHz, etc. Sample-and-hold circuits are used to sample and hold the oscillator signal values at the time of a pulse signal whose time is to be measured. Binary (twostate) signals are generated whose values correspond to the polarities of the sampled oscillator signal values. These binary signals are connected to simple logical circuits which resolve any ambiguities and generate the desired digital number in a normal binary or other desired form. This generated number is a digital measure of the time of occurrence of the pulse which initiated the sampling operation. Time intervals may be measured by two such sampling operations and a subtraction of the two resulting digital values.

Apparatus for electrically converting an analog signal into a digital representation

Abstract: A method is described for electrically converting an analog signal into a digital representation in a manner that maximizes noise rejection. The digital representation is formed from a preselected number of discrete points corresponding to sampled approximations of the analog signal. In establishing the magnitudes of the respective points, digital samples of the analog signal are taken at a predetermined number of discrete coordinates along the analog signal on either side of the respective discrete points. The predetermined number of coordinates are averaged and employed as corresponding approximations for the respective discrete points in the digital representative reproduction of the analog signal. The effects of harmonics of power line frequencies associated with processing electrical equipment are minimized by sampling the discrete coordinates for a particular point over an integral number of cycles of the power line frequency. In addition, noise having a high frequency, low duty cycle can be minimized by sampling a relatively large number of discrete coordinates over a period substantially greater than the occurrence of the noise.

Wideband signal transmission with digital to image signal conversion - is for transmission as pseudo TV signal with receiver end reconversion

Abstract: Wideband signal transmission is effected through conversion of the signal, which is digital or prepared by a digital process to the image signal of a pseudo-television signal in the transmitter. With superimposing of the synchronous pulses, transmission then takes place over a television channel. On the receiver side the synchronous signals are separated for control purposes. The image signal is then re-converted to the original digital signal form. The pseudo-television signal may be superposed in the region of the line blocking signals and synchronising pulses of the transmitter which then are separated at the receiver end before re-conversion of the image signal. On the transmitter side the signal source (SQ) is connected to a signal transformer (SSU).

Digital signal processing arrangement using a cascaded integrator function generator

Abstract: An analog-to-digital counting encoder or a digital-to-analog counting decoder for a pulse code modulation signal typically requires a function generator using precision components to generate a piecewise linear comparison signal corresponding to a segment companding law. As a result, nonproportional component change introduces nonlinear distortion in the encoded or decoded signal. To mitigate nonlinear distortion and to provide a less expensive digital signal processing arrangement, the invention resides in an improved nonprecision component function generator. The function generator includes two serially connected integrators which are advantageously switched in a complementary fashion to provide the comparison signal.

Digital Processing Of Synthetic Aperture Optical Imagery

Abstract: A synthetic-aperture approach offers one method for obtaining high resolution optical imagery with low-mass, satellite-based optical telescopes. Images recorded sequentially with different aperture configurations are sampled and digitized, and appropriate spatial-frequency-domain processing is performed on a computer to obtain the desired high-resolution imagery. Special steps can be introduced in these post-detection processing operations that allow a substantial reduction in the optical tolerances that must be maintained by the aperture elements in the imaging process.

Recent Progress in Digital Processing of Speech

Abstract: Current scientific efforts in the field of digital processing of speech are focused at the aims of improving the efficiency in the present state of the art, and of developing new digital speech communication systems. Therefore, thorough studies on the statistical characteristics of speech signals, speech coding, speech recognition, and speech synthesis are necessary. Recent results and actual trends are reviewed in this paper.

Implementation of delta modulators for digital inputs

Abstract: The delta modulation (DM) of signals which are already in digital form is of interest in a variety of signal-processing applications. It is shown that a delta modulator for digital inputs may be usefully interpreted as a process of slope bias, accumulation, and overflow detection leading to designs which are simpler than those based on analog techniques. Applications are discussed including a divider for delta-modulated signals and an averager of K delta-modulated inputs.