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Showing papers on "Digital signal processing published in 1979"


Book
01 Jan 1979
TL;DR: This text deals with the construction of algorithms that filter data into useful information and starts with the basics and goes on to cover advanced topics such as recursive and non-recursive filters (including optimization techniques), wave digital filters and DFTs.
Abstract: This text deals with the construction of algorithms that filter data into useful information. It starts with the basics and goes on to cover advanced topics such as recursive and non-recursive filters (including optimization techniques), wave digital filters and DFTs. A new chapter on the application of digital signal processing offers up-to-date techniques and there are new problems and examples throughout. Other features new to this second edition include chapters on quasi-Newton and minimax optimization algorithms for the design of recursive filters and equalizers, and efficient and robust algorithms for the design of non-recursive filters and differentiators. HLP computer language is now replaced with Pascal.

486 citations


Journal ArticleDOI
01 Jan 1979

310 citations



Journal ArticleDOI
S.S. Reddi1
TL;DR: Algorithms for locating multiple signal sources simultaneously with the aid of linear arrays and digital signal processing are presented and computer simulations are performed to investigate the behavior of the algorithms in terms of their accuracy, convergence, and angular resolution.
Abstract: Algorithms for locating multiple signal sources simultaneously with the aid of linear arrays and digital signal processing are presented. The approach basically consists of determining the eigenvalues/eigenvectors of a complex covariance matrix and solving a polynomial formed from the elements of eigenvectors. Computer simulations are performed to investigate the behavior of the algorithms in terms of their accuracy, convergence, and angular resolution. The performance of the algorithms with actual test data is reported.

222 citations


Book
01 Jun 1979

116 citations


Patent
19 Mar 1979
TL;DR: In this paper, an improved system for sensing and controlling the level of fluent material in two reservoirs when material is transferred from one reservoir to the other is presented. But the system is limited to the use of four sensors placed in each reservoir at desired "high" and "low" control levels.
Abstract: An improved system for sensing and controlling the level of fluent material in two reservoirs when material is transferred from one reservoir to the other. In electrically conducting materials, levels are sensed by direct current conduction between electrode pairs placed in each reservoir at desired "high" and "low" control levels which send D.C. digital logic sensing signals from these four sensors over connecting wires to four high resistance input circuits of a digital signal processing circuit; in electrically nonconducting materials, levels are sensed by suitable transducers which upon immersion send D.C. digital logic sensing signals to the digital signal processing circuit. The digital signal processing circuit receives the multiple sensor signals and provides a digital output signal or absence thereof, depending on the relative state and sequence of the sensing signals and transmits this digital output signal to the input circuit of a solid state relay, the output circuit of which controls the power of flow control means in a manner to maintain desired levels of fluent material in both reservoirs.

90 citations


Book
01 Nov 1979

73 citations


PatentDOI
TL;DR: A device to control machines by voice is disclosed which responds to a plurality of predetermined musical tones in a sequence to generate a digital control output signal.
Abstract: A device to control machines by voice is disclosed which responds to a plurality of predetermined musical tones in a sequence to generate a digital control output signal. Audio conversion means convert the sequence of tones into a corresponding sequence of digital number signals which are temporarily stored in a memory. Then a sequence of ratio signals are generated by dividing each of the digital number signals by one of the digital number signals. The resulting sequence of ratio signals is then converted into a corresponding sequence of digital numbers which comprises the digital control output signal. A microprocessor and associated storage processes the digital data and controls the conversion of the musical tones into corresponding digital numbers.

60 citations


Patent
28 Nov 1979
TL;DR: In this paper, a digital to analog converter is employed in the digital line circuit of a telephone system and operates to convert a digital signal indicative of an analog speech signal back into a replica of the analog signal.
Abstract: A digital to analog converter is employed in the digital line circuit of a telephone system and operates to convert a digital signal indicative of an analog speech signal back into a replica of the analog signal. The converter operates with an interpolated input digital signal to detect by means of a sign bit, the characteristic of an input digital word as being indicative of a positive or negative level. An error correcting signal is provided by the converter which is added to the next digital word to provide a compensated word having a sign bit determined by the remainder and the sign bit of the previous digital word. This word is then processed in sequence to produce an output pulse stream from the sign detector indicative of successive positive or negative values as defined by the input digital words, each of which are modified according to the error correcting signal.

55 citations



Book
01 Jun 1979
TL;DR: The author examines such topics as stability analysis of feedback control systems and digital filters subject to the effects of finite wordlength arithmetic; linear prediction, parameter identification, and relationships involving Kalman filtering and "fast" algorithms; system synthesis, realization, and implementation.
Abstract: A number of current research directions in the fields of digital signal processing and modern control and estimation theory were studied. Topics such as stability theory, linear prediction and parameter identification, system analysis and implementation, two-dimensional filtering, decentralized control and estimation, image processing, and nonlinear system theory were examined in order to uncover some of the basic similarities and differences in the goals, techniques, and philosophy of the two disciplines. An extensive bibliography is included.

Journal ArticleDOI
J. Justice1
TL;DR: This paper lays the foundations for a synthesis procedure based on the discrete Hilbert transform of harmonic structure in time for digital synthesis of waveforms.
Abstract: Digital synthesis of music has led to the consideration of models other than the usual additive (Fourier) synthesis of waveforms. One of these methods, based on the FM equation, has been found to be of particular value due to its easy implementation and the richness and evolutionary character of its harmonic structure in time. Any useful synthesis procedure should be accompanied by a corresponding analytic procedure. In this paper we lay the foundations for such a procedure based on the discrete Hilbert transform.

Journal ArticleDOI
A. Peterson1
01 Nov 1979

Journal ArticleDOI
01 Apr 1979
TL;DR: A review of signal processing methods which can be used to improve the effectiveness of systems designed for acoustic imaging and bearing estimation is presented, including stochastic acoustic signals, image processing, enhancement, and pattern recognition.
Abstract: A review of signal processing methods which can be used to improve the effectiveness of systems designed for acoustic imaging and bearing estimation is presented. Topics covered include a) signal processing for increased resolution, b) the processing of stochastic acoustic signals, c) image processing, enhancement, and pattern recognition. The discussion of resolution processing includes lateral resolution improvement by both superresolution techniques and aperture synthesis, and improvement of both range and Doppler resolution. The stochastic signal-processing section addresses adaptive processing, as well as methods of imaging in the case of incoherent, noisy signals.

Patent
11 May 1979
TL;DR: In this article, a data modem with 8-phase DPSK modulation is presented. But the modem includes a microprocessor and associated memories for digital signal processing and control of substantially all transmitter and receiver operations.
Abstract: A data modem is provided which is operative at a data rate of 4800 bps and employing 8 phase DPSK modulation. The modem includes a microprocessor and associated memories for digital signal processing and control of substantially all transmitter and receiver operations.

Patent
18 Dec 1979
TL;DR: In this paper, a symmetrical digital transversal filter was proposed to estimate the values of the omitted signals from the samples of the transmitted signal by making use of frequency components outside the normal defined bandwidth of the original signal.
Abstract: Encoded digital television signals having a defined bandwidth are transmitted at reduced bandwidth by regularly omitting (18) one in every n of the signal samples, where n is greater than two and is preferably three or four. At a receiver the omitted samples are regenerated by estimating (26) their values from the samples of the reduced-rate transmitted signal by making use of frequency components outside the normal defined bandwidth of the original signal. The estimation can be achieved by a symmetrical digital transversal filter of which one in every n coefficients is zero, the amplitude/frequency characteristics of the filter being unity throughout the defined bandwidth of the TV signal and antisymmetric about half the mean lower sample rate.

Journal ArticleDOI
TL;DR: FFT, one of the important tools of digital signal processing was used to filter surface profiles and this technique was compared with the existing methods available for filtering surface profiles.

Patent
21 Aug 1979
TL;DR: In this article, a monitor system consisting of an adder for summing up a plurality of unknown analog electric quantity signals denoting the operating conditions of such a transmission line, a multiplexer for generating output signals corresponding to the plural analog quantity signals and also to a signal denoted the result of addition made by said adder, an A-D converter for converting output analog signals from the multiple-xer into digital signals, and a monitor which sums up those of the output digital signals from a converter which correspond to said analog signals.
Abstract: A monitor system comprising: an adder for summing up a plurality of unknown analog electric quantity signals denoting the operating conditions of such as a transmission line; a multiplexer for generating output signals corresponding to the plural analog electric quantity signals and also to a signal denoting the result of addition made by said adder; an A-D converter for converting output analog signals from the multiplexer into digital signals; and a monitor which sums up those of the output digital signals from the A-D converter which correspond to said plural analog electric quantity signals, compares a signal denoting the result of said addition with a digital signal converted from an output signal from the adder, receives an error detection signal denoting the known prescribed electric quantity converted into a digital signal through the multiplexer and A-D converter, and judges the operating condition of the A-D converter from the contents of said received digital signal.

Patent
26 Dec 1979
TL;DR: In this article, a method and apparatus for providing mass storage of vocal information by means of a digital communication system for use with a training simulator is described, which includes an input device, such as a microphone, an encoder, and an output device for providing an audio or vocal output to a listener.
Abstract: A method and apparatus for providing mass storage of vocal information by means of a digital communication system for use with a training simulator is disclosed. Unlike prior art analog records and playback systems, such as tapes, etc., the inherent advantages and flexibility available with respect to a digital system including randomly accessible voice messages, convenience of editing the stored data, and management and control by the computational system of the training simulator are available with the present invention. In general, the system of this invention comprises an input device, such as a microphone (30), an encoder (54) for encoding the analog information from the microphone (32) to digital format, a digital processing controller or computer (20) for processing the digital communication data in response to a stored program such that it is suitable for use with the computational system, and a mass storage media (38) for storing the processed digital data. To provide audio response and playback from the stored digital data, the system includes a processor (36) for selectively reading stored bulk data and processing the selected data into a form suitable for decoding, a decoding means (66) for translating the digital data to analog information, and an output device (42) for providing an audio or vocal output to a listener.

Journal ArticleDOI
TL;DR: Three applications of digital signal processing in power system planning are discussed, including the use of a vector radix 2-D FFT in a spatial load growth model, which has reduced computation time for spatial convolutions by an order of magnitude.
Abstract: Three applications of digital signal processing in power system planning are discussed. The use of a vector radix 2-D FFT in a spatial load growth model has reduced computation time for spatial convolutions by an order of magnitude. The character of a power system is used to derive a set of 2-D digital filters that represent the power system's sensitivity to spatial load forecast-design errors. System design filters provide a useful way of analyzing long range power system needs.

Patent
19 Nov 1979
TL;DR: In this paper, a memory control system comprises a first memory supplied with an incoming modulated digital signal which is formed by subjecting an analog signal to digital signal processing of discontinuous level modulation system, and a first control circuit for supplying a control signal to the first memory.
Abstract: A memory control system comprises a first memory supplied with an incoming modulated digital signal which is formed by subjecting an analog signal to digital signal processing of discontinuous level modulation system, and a first control circuit for supplying a control signal to the first memory. The first control circuit producing the control signal for controlling the first memory in such a manner that the total memory capacity of the first memory is partitioned into a plurality (k) of memory capacity segments having given capacity values (lengths) for use, and the modulated digital signal is written in and further the modulated digital signal thus written in is read out with the order thereof rearranged, interrelatedly with the circulation of addresses through the plurality of divided memory capacity segments while maintaining constant the relationship in terms of capacity values (lengths) between the plurality of divided memory capacity segments.

Journal ArticleDOI
TL;DR: New algorithms to self-deconvolve such experimental spectra as those obtained in AES and APS are devised, based upon the fast Fourier transform algorithm and digital signal processing concepts.

Patent
20 Jun 1979
TL;DR: In this paper, a digital amplitude control apparatus for a digital audio signal is proposed, in which the output voltage of a linear-type potentiometer or attenuator connected across a DC power supply is converted into a digital signal by an A/D converter, the output signal being linearly related to the manipulated variable of the slider of the potentiometers, and the digital signal is applied as an address signal to a memory which stores in memory locations thereof a series of amplitude control digital values with a desired amplitude changing characteristic.
Abstract: A digital amplitude control apparatus for a digital audio signal in which the output voltage of a linear-type potentiometer or attenuator connected across a DC power supply is converted into a digital signal by an A/D converter, the output voltage being linearly related to the manipulated variable of the slider of the potentiometer, the digital signal is applied as an address signal to a memory which stores in memory locations thereof a series of amplitude control digital values with a desired amplitude changing characteristic, thereby reading out of an accessed location a amplitude control value corresponding to the attenuation set by the potentiometer, and a digitized audio information signal to be amplitude-controlled is multiplied by the amplitude control value read out of the memory by means of a digital multiplier.

Proceedings ArticleDOI
01 Jan 1979
TL;DR: Low power, standard CMOS/SOS LSI technology, applied to a 6b (expandable to 7b) video speed A/D converter design, will be reported.
Abstract: Low power, standard CMOS/SOS LSI technology, applied to a 6b (expandable to 7b) video speed A/D converter design, will be reported. Applications and potential VLSI digital signal processing systems will be reviewed.

Dissertation
18 Jun 1979
TL;DR: The design of the encoding system and specifications of system parameters are developed from the perceptual requirements and digital signal processing techniques, and the system is designed to exploit the limited detection ability of the auditory system.
Abstract: : The development of a digital encoding system for speech and audio signals is described. The system is designed to exploit the limited detection ability of the auditory system. Existing digital encoders are examined. Relevant psychoacoustic experiments are reviewed. Where the literature is lacking, a simple masking experiment is performed and the results reported. The design of the encoding system and specifications of system parameters are then developed from the perceptual requirements and digital signal processing techniques. The encoder is a multi-channel system, each channel approximately of critical bandwidth. The input signal is filtered via the quadrature mirror filter technique. An extensive development of this technique is presented. Channels are quantized with an adaptive PCM scheme. The encoder is evaluated for speech and audio signal inputs. For 4.1-kHz bandwidth speech, the differential threshold of encoding degradation occurs at a bit rate of 34.4 kbps. At 16 kbps, the encoder produces toll-quality speech output. Audio signals of 15-kHz bandwidth can be encoded at 123.8 kbps without audible degradation.

Journal ArticleDOI
TL;DR: The authors describe high-performance CMOS LSIs for digital signal-processing (DSP) technology, such as digital filter, fast Fourier transform (FFT), discrete Fouriertransform (DFT), and digital phase-locked loop (DPLL), for communication use.
Abstract: The authors describe high-performance CMOS LSIs for digital signal-processing (DSP) technology, such as digital filter, fast Fourier transform (FFT), discrete Fourier transform (DFT), and digital phase-locked loop (DPLL), for communication use. Device design for high-speed and low-power CMOS is described and its feasibility is shown as characteristics of propagation delay time and power delay product.

PatentDOI
TL;DR: A voice detector for supplying a control signal in an automatic telephone answering apparatus in response to a voice signal received from a telephone line.
Abstract: A voice detector for supplying a control signal in an automatic telephone answering apparatus in response to a voice signal received from a telephone line. A first one of two parallel signal channels generates a first digital signal for an incoming signal having a minimum amplitude with a minimum duration. The second signal channel generates a second digital signal for an incoming signal having a minimum number of consecutive cycles with periods within a predetermined range corresponding to the frequency range of a ringing signal. A logic element combines the first and second digital signals to produce the control signal if simultaneously the first digital signal is present and the second digital signal is absent. Thereby, the control signal is produced only in response to an incoming voice signal, but not in response to a ringing or a noise signal.

Journal ArticleDOI
TL;DR: This paper describes high-performance CMOS LSI's for digital signal-processing (DSP) technology, such as digital filter, fast Fourier transform (FFT), discrete Fouriertransform (DFT), and digital phase-locked loop (DPLL).
Abstract: This paper describes high-performance CMOS LSI's for digital signal-processing (DSP) technology, such as digital filter, fast Fourier transform (FFT), discrete Fourier transform (DFT), and digital phase-locked loop (DPLL). First, DSP functions for communication use, functional blocks to compose DSP functions, and the types of arithmetic for LSI are discussed. It is explained that multiplier (MPL), variable-length shift register (VSR), and linear arithmetic processor (LAP) have been chosen as the most useful DSP LSI's. Device design for high-speed and low-power CMOS is described and its feasibility is shown as characteristics of propagation delay time at 430 ps and power delay product at 0.073 pJ. The 3-µm effective channel-length CMOS technology has been selected for the DSP LSI because of the high speed, 5 ns, in the case of two input NAND gates and high yield technology. The multiplier architecture is pipeline and uses the Two's-complement representative, the variable-length shift register uses the binary-select method, and the linear arithmetic processor uses the method of changing the outside connections for realization of DSP functions. Maximum operating frequency of these LSI's is more than 23 MHz at the 5-V source voltage. Power dissipation of a VSR, which has been lossy, is less than 250 mW in the 8-MHz operation. They have wider application to communication systems. High-speed CMOS technology is applied to the digital system equipment up to the second level of the PCM hierarchy.

Journal ArticleDOI
TL;DR: An inline holographic system suitable for long-wavelength holography is studied by computer simulation and a digital processing technique for the reduction of the interfering outputs in the reconstructed image is given.
Abstract: An inline holographic system suitable for long-wavelength holography is studied by computer simulation. A digital processing technique for the reduction of the interfering outputs in the reconstructed image is given. Results arc compared with those obtained from direct phase and amplitude recording.

Patent
Harold Henry Harris1
17 Sep 1979
TL;DR: In this article, a line unit interfaces voice band signals and wide band signals extending from D.C. into the voice band between an analog line and a digital telecommunication facility.
Abstract: A line unit interfaces voice band signals and wide band signals extending from D.C. into the voice band between an analog line and a digital telecommunication facility. The line unit includes an analog to digital converter, a control circuit for generating one of two control signals and a line circuit. The line circuit includes a hybrid circuit in a first signal path for carrying voice band analog signals, and a second signal path for carrying wide band analog signals from the analog line. The signals in either signal path are transmitted to the analog to digital converter for encoding by way of a first switch or a second switch connected in series with the first and second signal paths respectively. The first and second switches are alternately conductive in response to one or the other of the control signals respectively.