Showing papers on "Digital signal processing published in 1976"

Orthogonal transforms for digital signal processing

Abstract: A tutorial-review paper on discrete orthogonal transforms and their applications in digital signal and image (both monochrome and color) processing is presented. Various transforms such as discrete Fourier, discrete cosine, Walsh-Hadamard, slant, Haar, discrete linear basis, Hadamard-Haar, rapid, lower triangular, generalized Haar, slant Haar and Karhunen-Loeve are defined and developed. Pertinent properties of these transforms such as power spectra, cyclic and dyadic convolution and correlation are outlined. Efficient algorithms for fast implementation of these transforms based on matrix partitioning or matrix factoring are presented. The application of these transforms in speech and image processing, spectral analysis, digital filtering (linear, nonlinear, optimal and suboptimal), nonlinear systems analysis, spectrography, digital holography, industrial testing, spectrometric imaging, feature selection, and patter recognition is presented. The utility and effectiveness of these transforms are evaluated in terms of some standard performance criteria such as computational complexity, variance distribution, mean-square error, correlated rms error, rate distortion, data compression, classification error, and digital hardware realization.

On the hardware implementation of digital signal processors

Abstract: An approach to the machine organization of dedicated hardware digital signal processors is proposed that is based on a specialized representation of the processing coefficients derived from the canonical signed-digit code. This leads to a realization requiting the minimum number of add/subtract operations to mechanize the required multiplications and additions. The proposed organization is shown to be highly modular and well suited to integrated circuit implementation, and offers a significantly better performance when compared with existing realizations using prepackaged multipliers.

Audio storage and distribution system

Abstract: This invention relates to an electronic system and a method for storing and distributing audio signals over existing communication lines. The system comprises a compressor for compressing in a predetermined manner the waveform amplitude of an input analog signal, thereby forming a compressed analog signal. The compressed analog signal is then converted into a digital signal by an analog to digital converter. A digital interface subsystem stores and retrieves selected ones of the digital signals for transmission over a communications line. At a remote end of the communications line the digital signal is converted back to its analog compressed signal representation by a digital to analog converter. The compressed analog signal is then expanded in a manner complimentary to the compressor operation, thus reconstructing the analog signal. A selector generator is provided at the remote end of the communications line for generating a command signal over the communications line to command the digital interface subsystem to select the desired one of the stored digital signals.

Second order motion compensator for high resolution radar

Abstract: A method and apparatus for providing motion compensation that allows dynamically changing flight paths during high resolution, squinted, synthetic aperture mapping by making use of a second order motion compensation by means of a two-stage correlator configuration utilizing digital signal processing techniques.

A pitch-synchronous digital feature extraction system for phonemic recognition of speech

Abstract: The system described in this paper is subdivided into three main steps: pitch extraction, segmentation, and formant analysis. The pitch extractor uses an adaptive digital filter in time-domain transforming the speech signal into a signal similar to the glottal waveform. Using the levels of the speech signal and the differenced signal as parameters in time domain, the subsequent segmentation algorithm derives a signal parameter which describes the speed of articulatory movement. From this, the signal is divided into "stationary" and "'transitional" segments; one stationary segment is associated to one phoneme. For the formant tracking procedure, a subset of the pitch periods is selected by the segmentation algorithm and is transformed into frequency domain. The formant tracking algorithm uses a maximum detection strategy and continuity criteria for adjacent spectra. After this step, the total parameter set is offered to an adaptive universal pattern classifier which is trained by selected material before working. For stationary phonemes, the recognition rate is about 85 percent when training material and test material are uttered by the same speaker. The recognition rate is increased to about 90 percent when segmentation results are used.

A digital system to produce imagery from SAR data

Abstract: This paper describes a digital processing algorithm and its associated system design for producing images from Synthetic Aperture Radar (SAR) data. The proposed system uses the Fast Fourier Transform (FFT) approach to perform the two-dimensional correlation process. The range migration problem, which is often a major obstacle to efficient processing, can be alleviated by approximating the locus of echoes from a point target by several linear segments. SAR data corresponding to each segment is correlated separately, and the results are coherently summed to produce full-resolution images. This processing approach exhibits greatly improved computation efficiency relative to conventional digital processing methods.

Stereophonic television sound transmission system

Abstract: A compatable stereophonic television sound transmission system for transmitting left and right audio signals in conjunction with a television broadcast wherein video information is conveyed on an amplitude-modulated carrier in a frequency channel having defined frequency limits. The system includes at a transmitter location a multiplex generator for generating a composite signal having a first component representative of the sum of the audio signals, a second component comprising an amplitude-modulated suppressed carrier subcarrier signal representative of the difference between the audio signals, and a pilot component representative of the phase and frequency of the suppressed carrier. The composite signal is utilized to frequency-modulate a sound carrier to develop an RF signal component which is added to the television channel at a discrete frequency spacing from the video carrier. The system includes at a receiver location a tuner for converting the transmission channel to an intermediate frequency, a filter for separating the sound signal therefrom, and a detector for deriving the composite signal from the sound signal. The composite signal is demodulated in a stereo demodulating stage to develop the left and right audio signals. Improved noise performance is obtained in the system by applying preemphasis to the left and right audio signals, preemphasis of the composite signal, enhancement of the second composite signal component, and/or Dolby-B processing of the left and right audio signals or to the composite signal. An adapter is shown for utilizing the system for bilingual programming.

A correct proof of Huang's theorem on stability

Abstract: A correct proof of Huang's theorem on the stability of two-dimensional causal recursive digital filters is developed using a maximum modulus theorem for algebraic functions.

Digital controller system

Abstract: A digital controller system for controlling the flow of a plurality of processes, each of the processes being provided with a process detector and an actuator, the digital controller system comprising: a digital bus for transmitting digital signals from one location to another; an analog bus for transmitting analog signals from one location to another; a plurality of direct digital loop stations, each of said direct digital loop station adapted to be connected to a process detector and an actuator of a process for receiving analog signals characterizing the state of the process from the process detector and transmitting the analog signals onto the analog bus and for receiving digital signals controlling the flow of the process from the digital bus and transmitting corresponding analog signals to the operating apparatus; and a central processor unit for transmitting the digital signals controlling the flow of the process to the digital bus, and for receiving the analog signals characterizing the state of the process from the analog bus, the central processor unit including a digital computer, an analog-to-digital converter and a data transmission unit.

A Digital Receiver for Tone Detection Applications

Abstract: A digital circuit suitable for detection of tones in signaling applications is described. The amount of hardware required for the realization of the circuit is shown to be quite small. The circuit may be used for both analog and digital input signals. For analog signals, the necessary A/D conversion becomes very simple. Results of simulations on a digital computer are given that indicate the good performance of the circuit.

Charge-coupled devices for analog signal processing

Abstract: CCD's are inherently analog and can be used to implement a number of sampled data filtering functions in the analog domain. The CCD transversal filter is a particularly cost effective building block because of its versatility and simplicity. The limitations and applications of CCD transversal filters are summarized, and their cost advantages relative to digital filters are discussed.

Code tracking signal processing system

Abstract: A code tracking signal processing system for tracking a coded signal of the pseudo-random-noise, or PRN, type wherein a time shift comparison is made of the input coded signal and a pair of time estimated coded feedback signals which represent an estimate of the coded input signal which has been advanced and delayed, respectively, by the same specified time shift. An effective error signal is formed from the time shift comparison signals and supplied to a digital integration means, such as an up-down counter, to generate a pair of control signals. The control signals control the pulse rate of a pulsed clock signal in accordance with the time shift error between the estimated code signal and the input code signal. The controlled pulse rate signal is then used to generate the advanced and delayed feedback signals and to produce a coded signal which is in effect locked into time synchronism with the input coded signal.

'Straightening through smoothings' seen as a digital filter

Abstract: It is shown that the digital signal processing method called 'straightening through smoothings' is exactly equivalent to a digital filter, even including end-effect corrections.

Digital control system for motors

Abstract: A phase-control type of closed-loop control system for motors. A reference signal generator generates a digital reference signal having a binary value which varies between first and second values n times during each a-c input voltage cycle. An error signal generator generates a digital error signal having a binary value which varies in accordance with the difference between the desired and actual values of a motor variable such as armature current. The most significant bits of the digital error signal are utilized to select the angular range within which firing signals will be applied to a phase-controlled rectifier network that is arranged to drive the motor. The least significant bits of the digital error signal are compared with the digital reference signal to determine the time, within the selected angular range, when firing signals will be applied to the controlled rectifier network. By use of this technique, the control system provides both precisely timed control events and a range of control which extends over the entire motoring and inverting regions of motor operation.

Adaptive correction of phase errors in noncoherent demodulation of carrier asymmetrically modulated with digital signals

Abstract: Digital signals asymmetrically modulated upon a carrier, with suppressed or vestigial second sideband, are recovered at a receiver by pseudo-coherent demodulation or periodic sampling and subsequent digitization to provide a train of raw data signals X from which a train of raw quadrature signals Y is derived by digital filtration. Signals X and Y are fed to a phase corrector where they are cross-multiplied with a sine function and a cosine function of a feedback signal W, approximating a corrective phase angle φ(t), to yield a corrected in-phase signal X' and a corrected quadrature signal Y'. Signal X' is quantized to provide a reference signal c which, upon subtraction from signal X', produces a bipolar difference signal whose sign bit is multiplied with either the entire signal Y' or its sign bit to provide an error signal V. The latter is averaged over a number of clock cycles, resulting in the feedback signal W whose trigonometric functions are read out from a memory for utilization in the generation of corrected signals X' and Y'.

Digital PSK modem

Abstract: In the transmitter portion a multiplexer multiplexes first digital data and first teletype signals to provide first digital data signals. The first digital data signals are converted into second digital data signals having a different form than the first digital data signals by a Viterbi coder. A PSK modulator responsive to the second digital data signals provides a first PSK signal for transmission. In the receiver portion a PSK demodulator receives a second PSK signal containing third digital data signals having multiplexed second digital data and second teletype signals with the third digital data signals being extracted from the second PSK signal. A Viterbi decoder converts the third digital data signals into fourth digital data signlas having a different digital form than the third digital data signals. A PSK demultiplexer is responsive to the fourth digital data signals to separate the second digital data and the second teletype signals for utilization. The arrangement to extract the third digital data signals from the second PSK signals includes the PSK demodulator and a Costas' type phase locked loop including a number controlled oscillator, a digital phase shifter, an accumulator arrangement including three accumulators accumulating different data and errors, a clock recovery circuit and a loop filter controlling the number controlled oscillator.

Analog to digital converter having nonlinear amplitude transformation

Abstract: An analog to digital converter simultaneously digitizes a gamma corrected analog video signal and removes the gamma correction. The converter has a plurality of parallel analog comparators which compare the video signal amplitude with the amplitude of a plurality of different reference signals, each comparator providing an output representative of the respective comparison result. Each reference signal amplitude is a different multiple of a fraction, of a common reference signal, raised to the gamma power thereby removing the gamma correction. A decoder responsive to the output of the comparators generates the digital output signal in a preselected digital code.

Circuit for the conversion of a digital signal to an analog signal

Abstract: A digital-to-analog converter using a combined pulse rate and pulse width modulation of an output signal of a circuit for providing a digital signal to be converted to a periodically occurring series of digital comparison signals to be obtained by a proper choice of a particular series. The digital-to-analog converter has a very small error which, due to its low temperature sensitivity, is particularly suitable for tuning circuits of television receivers.

Automatic generation of time efficient digital signal processing software

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.

Digital acquisition and interactive processing of ultrasonic echoes

Abstract: The requirements for an interactive digital signal processing system for ultrasonic pulse-echo signals are discussed. A system based on an Interdata Model 80 mini-computer and micro-processor interface is described. The system is capable of acquiring ultrasonic data at a sampling rate of 6 MHz. Ultrasonic B-mode data may be acquired in Line Mode, when echo waveform data and transducer position and orientation are stored, or in Section Mode when the data is converted directly into picture form in memory in the same way that a standard echogram is formed on the screen of an oscilloscope. In each case the data for single complete high resolution echogram may be acquired in less than 15 sec. It is shown that the 6 MHz sampling rate is sufficient to faithfully preserve the echo waveshape of a 2 MHz system independently of the relation to the phase of the sampling. Also shown is a cross-sectional echogram of the pregnant uterus, and its digital representation with a raster density of 80 × 100 and 160 × 200 picture elements. The computer is programmed with an interactive program to allow ultrasonic signals to be acquired, stored, processed and examined with the convenience of a desk calculator. Sample operations are illustrated including data interpolation, spectrum analysis, filtering and complex signal deconvolution. The ability of deconvolution techniques to resolve targets separated by less than one wavelength in depth is demonstrated. Possibilities of further processing techniques are outlined.

Techniques for the digital interfacing of t.d.m.-f.d.m. systems

Abstract: The problems of digital interfacing of t.d.m.—f.d.m. systems are considered. A novel method of converting t.d.m. to f.d.m. signals using digital signal processors is proposed. This method makes use of a periodically varying digital filter coupled with a fast Fourier transformer. The combination of these two devices takes advantage of the lowest possible sampling rate for the system, and thus greatly reduce the complexity and cost of the convertor. The method can be applied equally well for f.d.m. to t.d.m. conversion.

Properties of Low Precision Analog-to-Digital Converters

Abstract: The problem of quantization and saturation noise introduced by the process of analog-to-digital conversion is addressed. Analog-to-digital converters (ADC) with even versus odd numbers of output states are compared. Expressions are derived and evaluated which yield the signal-to-noise ratio and the gain versus signal level input when the input signal has an assumed Gaussian probability density. The results presented should have application in all fields in which digital signal processing is performed.