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Showing papers on "Digital signal processing published in 1968"


Journal ArticleDOI
TL;DR: The digital filter with complex coefficients finds applications in the digital processing of analytic signals and complex envelopes and a theory is developed for designing such filters based on low-pass analog prototypes and digital design techniques for real filters.
Abstract: The digital filter with complex coefficients finds applications in the digital processing of analytic signals and complex envelopes. A theory is developed for designing such filters based on low-pass analog prototypes and digital design techniques for real filters. An example of a filter designed according to this theory is presented. The relative advantages of real and analytic signal processing are discussed. It is shown that the filtering required by either processing technique requires essentially the same amount of signal operations, which is reasonable in view of the fact that the same amount of information is processed in both classes of filter.

72 citations



Journal ArticleDOI
TL;DR: The design procedure of optimum digital filters for analog-digital phase-locked loops is given and it is shown that the optimum digitalanalog phase- Locked loop is not the discrete version of the optimum continuous phase-lock loop.
Abstract: This paper gives the design procedure of optimum digital filters for analog-digital phase-locked loops. The inputs considered are the step and ramp change in phase. The digital filters can easily be realized on the digital computer or otherwise. Design curves are given to choose proper noise bandwidth, sampling period, and loop parameters. It is shown that the optimum digitalanalog phase-locked loop is not the discrete version of the optimum continuous phase-locked loop.

32 citations


Journal ArticleDOI
TL;DR: A digital processor capable of computing the discrete Fourier transform for a range of audio signals in real time has been built as part of a facility to conduct research in signal processing.
Abstract: —A digital processor capable of computing the discrete Fourier transform for a range of audio signals in real time has been built as part of a facility to conduct research in signal processing. The digitized sample values can be complex. The arithmetic unit is configured to perform complex connectives, and automatic array scaling is used to make numerical accuracy independent of signal level. The Cooley–Tukey "fast Fourier transform" is the algorithm used.

25 citations


Journal ArticleDOI
TL;DR: A new data-transmission method that has twice the speed capability of the duobinary technique over an identical bandpass transmission medium is presented, which is sufficiently general to be applicable to both voice anti broadband communication channels.
Abstract: A new data-transmission method that has twice the speed capability of the duobinary technique over an identical bandpass transmission medium is presented. Advantages of this method include promising performance and completely digital manner of signal generation, thus considerably reducing the circuit complexity. The system is based on correlative techniques in which the digital encoding forms an integral part of the carrier modulation. The process is sufficiently general to be applicable to both voice anti broadband communication channels.

16 citations


Journal ArticleDOI
TL;DR: This paper attempts to set the stage for the companion papers on digital filtering to follow in this topical issue by introducing the z-transform of a discrete-time series and the use of this transform in linear system analysis.
Abstract: Digital filtering is the process of spectrum shaping of signal waveforms, using digital components as the basic dements for implementation This process is extensively used in the computer simulation of analog filters The unmistakable trends toward increased speed and decreased cost and size of digital components make digital filtering especially attractive at this time These trends promise to end the virtual monopoly of analog components for realizing real-time filters This paper attempts to set the stage for the companion papers on digital filtering to follow in this topical issue After introducing the z-transform of a discrete-time series, the use of this transform in linear system analysis is considered The relationship between discrete and continuous signals and systems is then discussed Since all the papers of this issue are concerned with digital filter implementations in one form or another, only an overview of these implementations is given here These include filter configurations, design methods, quantization effects, and the fast convolution method for implementing nonrecursive filters

16 citations


Patent
21 Nov 1968
TL;DR: In this article, the authors proposed a method for converting binary coded digital signals into analog signals, in which the initial digital signal is at first converted into a pulse whose duration is representative of said digital signal, then integrated.
Abstract: Apparatus for converting binary coded digital signals into analog signals, in which the initial digital signal is at first converted into a pulse whose duration is representative of said digital signal, then integrated. Also disclosed are some improvements allowing harmonics to be reduced, more particularly by combining a first pulse whose duration is representative of the binary value of the digital signal with a second pulse whose duration is representative of the binary complement of the value of said digital signal.

12 citations


Journal ArticleDOI
TL;DR: Only 15 minutes of programmed training in listening to this processed speech was found to be significantly effective for improving perception in all four conditions, and the effect of a short training period on the perception of the speech signals so processed.
Abstract: The purpose of this study was 1) to measure the intelligibility of speech signals processed with a frequency converter designed to effect a spectral compression by a sampling-synthesizing technique, and 2) to investigate the effect of a short training period on the perception of the speech signals so processed. The input signals were composed of 71 CVC words recorded by an American female speaker. The highest center frequency for the input of the 22 channels was 9500 Hz. The four compression ratios of the output to the fixed input frequency spectrum were 1.0, 0.7 0.5, and 0.4 for all 22 channels used. Eighty listeners were randomly selected and assigned to one of the four conditions of spectral compression. The intelligibility score of each condition of spectral compression was obtained from a multiple choice test of six possible choices for each stimulus. The average intelligibility score for all English phonemes was obtained for each condition of compression. The intelligibility of each vowel and consonant was also calculated for all conditions. Only 15 minutes of programmed training in listening to this processed speech was found to be significantly effective for improving perception in all four conditions.

7 citations


Journal ArticleDOI
TL;DR: The use of RADA techniques with efficient encodingdecoding and a processing repeater can provide a multiple access capability which efficiently uses bandwidth and downlink power.
Abstract: This paper describes a signal processing repeater which enhances the performance that can be obtained with random access discrete address (RADA) digital transmission links which operate in a multiple access mode. In general, RADA modulation techniques employ groups of pulses to transmit information. The pulses within a group normally have different carrier frequencies and are separated from each other by one or more time slots. The communication performance capability of a system which uses these techniques is a function of interference between the pulse groups of different users and background noise. A number of RADA techniques were proposed for the U. S. Army to provide communications within field army and divisions. For army application these had the advantages of rapid synchronization and ease of implementation, accompanied by the disadvantage of requiring large amounts of bandwidth. Significant bandwidth reductions can be obtained through the use of efficient data encoding-decoding techniques. Also, the downlink power requirements can be minimized through the use of an antipodal downlink signal. The processing repeater discussed in this paper can convert any number of randomly phased uplink signals to a single antipodal downlink signal which is amenable to phase coherent detection. The required downlink power is thereby minimized. Thus, the use of RADA techniques with efficient encodingdecoding and a processing repeater can provide a multiple access capability which efficiently uses bandwidth and downlink power.

5 citations


Journal ArticleDOI
TL;DR: Three methods for time domain design of finite-memory digital filters are presented and these filters can perform the operations of smoothing, prediction, and differentiation either separately or simultaneously.

5 citations



Journal ArticleDOI
TL;DR: A telemetry data processing system based on crosscorrelation techniques and maximum-likelihood detection principles and designed to decode a special type of pulse-code modulated (PCM) signals, pulse-frequency modulation (PFM), is discussed.
Abstract: A telemetry data processing system based on crosscorrelation techniques and maximum-likelihood detection principles and designed to decode a special type of pulse-code modulated (PCM) signals, pulse-frequency modulation (PFM), is discussed. The decoding system employs a high-speed multiple-pass tape scanner to sequentially generate a set of correlation indices for each data word using the alphabet of possible data words. An on-line digital computer functions as the decision device to select the greatest of the indices as representing the transmitted data word. The data processing system combines analog-digital hybrid techniques for relatively high-speed signal processing and yet requires only a small digital computer memory.

01 Jan 1968
TL;DR: The digital filter with complex coefficients finds applications in the digital processing of analytic signals and complex envelopes and a theory is developed for designing such filters based on low-pass analog prototypes and digital design techniques for real filters.
Abstract: The digital filter with complex coefficients finds applications in the digital processing of analytic signals and complex envelopes. A theory is developed for designing such filters based on low-pass analog prototypes and digital design techniques for real filters. An example of a filter designed according to this theory is presented. The relative advantages of real and analytic signal processing are discussed. It is shown that the filtering required by either processing technique requires essentially the same amount of signal operations, which is reasonable in view of the fact that the same amount of information is processed in both classes of filter.

01 Jan 1968
TL;DR: The final author version and the galley proof are versions of the publication after peer review that features the final layout of the paper including the volume, issue and page numbers.
Abstract: • A submitted manuscript is the version of the article upon submission and before peer-review. There can be important differences between the submitted version and the official published version of record. People interested in the research are advised to contact the author for the final version of the publication, or visit the DOI to the publisher's website. • The final author version and the galley proof are versions of the publication after peer review. • The final published version features the final layout of the paper including the volume, issue and page numbers.

30 Jun 1968
TL;DR: Adaptive array processing, dynamic programming, digital data transmission, recursive adaptive equalizers, and finite memory communication systems are discussed in this article, where adaptive array processing and dynamic programming are combined.
Abstract: Adaptive array processing, dynamic programming, digital data transmission, recursive adaptive equalizers, and finite memory communication systems