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Communications, IEEE Transactions on

Issue 4 • Date April 1984

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Displaying Results 1 - 24 of 24
  • [Front cover and table of contents]

    Publication Year: 1984 , Page(s): 0
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    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1984 , Page(s): 0
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    Freely Available from IEEE
  • A Minimum Dispersion Combiner for High Capacity Digital Microwave Radio

    Publication Year: 1984 , Page(s): 419 - 428
    Cited by:  Papers (6)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (856 KB)  

    A new minimum-dispersion (MID) combiner, which reduces multipath degradation in a high capacity digital microwave radio, is proposed. A high capacity digital microwave radio is inherently very sensitive to waveform distortion caused by multipath in-band delay dispersion and in-band amplitude dispersion. To minimize the in-band dispersion, the combined-signal spectrum shape from the two antennas is monitored before and after a small change takes place in the combining phase. The phase shifter rotates in the direction of the flatter of the spectrum shapes, either the one before or the one after monitoring. Performance evaluations through simulation calculation and theoretical estimation using in-band amplitude dispersion probability density are given. More improvement can be obtained when the MID combiner is used instead of the maximum power (MAP) combiner currently in use. Laboratory and field experiments, using a 200 Mbit/ s 16-QAM signal, verify these analyses and show an additional outage reduction factor of more than 5. View full abstract»

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  • A Low-Cost Digital Matched Filter for Arbitrary Constant-Envelope Spread-Spectrum Waveforms

    Publication Year: 1984 , Page(s): 354 - 361
    Cited by:  Papers (14)  |  Patents (23)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (768 KB)  

    In this paper a noncoherent digital matched filter (DMF) with binary quantization is presented which shows good performance for arbitrarily angle modulated constant-envelope waveforms having large time-bandwidth products. The considered DMF represents an extension of the noncoherent DMF with one-bit digitization suggested previously by Turin for binary PSK signals. Analysis of the filter performance in the presence of white Gaussian noise results in a general expression which allows the output SNR of both DMF's to be calculated for any spread-spectrum waveform. Evaluating this expression reveals that the extended DMF can be matched to arbitrary waveforms having constant envelopes in such a way that the SNR loss with reference to the corresponding analog matched filter is less than 2.87 dB. The degradation is as low as 1.96 dB for waveforms with biphase or quadriphase signal samples such as binary PN-PSK, PNQPSK, and offset-PN-QPSK, but also PN-MSK. If both DMF's are applied to waveforms having complex valued envelopes, the extended DMF may outperform Turin's DMF by as much as 3 dB. View full abstract»

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  • Modulation Transfer Noise Effects from a Continuous Digital Carrier to FDM/FM Carriers in Memoryless Nonlinear Devices

    Publication Year: 1984 , Page(s): 337 - 345
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (760 KB)  

    An analysis of the modulation transfer noise effects from a continuous (nonbursty) digital carrier to multiple FDM/FM carriers in a common memoryless nonlinear amplifier (e.g., satellite transponder TWTA) is presented. It is shown that the modulation transfer noise, which is caused by the random envelope variations of the digital carrier due to the random data patterns and filtering of the PSK pulses, consists of both discrete and continuous spectral components in the baseband of the FDM/FM carriers. Numerical examples in terms of the noise power ratio as well as a comparison between measured and computed results are given. In general, the discrete or tonal interference component at the symbol rate of the digital carrier is quite significant. Results of this paper should be useful in the planning of FDMA satellite transponders employing the mixed PSK-FM mode of operation. View full abstract»

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  • A New Frequency Domain Speech Scrambling System Which Does Not Require Frame Synchronization

    Publication Year: 1984 , Page(s): 444 - 456
    Cited by:  Papers (14)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1888 KB)  

    Communication security is becoming more and more important today. There is thus an increasing interest in analog scramblers due to the desire for secure speech communications over existing telephone channels with standard telephone bandwidth at acceptable speech quality and reasonable cost. The concept of scrambling the sample values of the speech waveforms becomes attractive due to its higher degree of security compared to the traditional scramblers. But all these sample value scramblers require frame synchronization, i.e., the signal segments used in scrambling and descrambling processes have to be exactly the same for signal recovery. This complicates the implementation and makes the transmission very sensitive to channel conditions. In this paper, a new frequency domain scrambling algorithm is presented, which is an extension of the DFT scrambler previously proposed. The use of short-time Fourier analyis and the filter bank concept leads to the special feature that frame synchronization is completely unnecessary. This simplifies the implementation and improves the reliability and feasibility. The theoretical developments, simulation results, hardware implementation, and test results are all discussed in detail. View full abstract»

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  • The Combined Effects of Noise and Multipath Propagation in Multilevel PSK Radio Links

    Publication Year: 1984 , Page(s): 411 - 418
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (720 KB)  

    A general analysis of the behavior of multilevel PSK radio links is carried out under multipath propagation conditions. A method is proposed to evaluate system unavailability due to the combined effects of noise and faded channel distortion. The "outage domain," having fixed the transmission channel, is calculated in the space of the variables which characterize any anomalous propagation condition. In particular, with reference to a "twopath model" (a "direct ray" and an "echo"), variations of carrier and symbol synchronisms due to the echo are taken into account and their importance is emphasized in realizing the numerical results. These are shown to be in very good agreement with the experimental ones which have appeared in the literature. An analytical realization of the results obtained is developed and leads to a simple asymptotic formula explaining outage probability as a function of some parameters of the echo statistics, the transmission channel adopted, and the number of modulation levels. Rapid evaluation of this formula allows us to plot outage probability as a function of spectral efficiency, measured in bits/s/Hz. View full abstract»

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  • Mutual Tracking in Phase and Delay of Randomly Modulated Signals

    Publication Year: 1984 , Page(s): 362 - 371
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (768 KB)  

    A two-parameter tracking loop that mutually locks the phases and delays of two noise-contaminated replicas of a randomly phase-modulated signal, received via different channels, is analyzed. It is shown that, at high SNR, i.e., small tracking errors, the twoparameter loop can be considered as two uncoupled single parameter loops, provided the signals are beaten down to zero center frequency. Special emphasis is put on the delay error variance; it is found that, in general, the broadening of the phase loop bandwidth (provided for more rapid acquisition) induces larger delay-tracking errors. Also, such broadening decreases the expected time to loss of delay lock; the influence of the location of the absorbing boundaries for this problem is investigated. Quantitative results for a typical configuration are presented in Figs. 1-9. View full abstract»

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  • The Information-Bit Error Rate for Block Codes

    Publication Year: 1984 , Page(s): 474 - 476
    Cited by:  Papers (32)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (288 KB)  

    The relations between the word error probability and the decoding algorithms for block codes are reviewed. A simple approximation that does not depend upon the code weight structure or the decoding details is derived for the information-bit error rate in terms of the channel-symbol error probability. View full abstract»

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  • Simultaneous Adaptive Estimation of Carrier Phase, Symbol Timing, and Data for a 49-QPRS DFE Radio Receiver

    Publication Year: 1984 , Page(s): 429 - 443
    Cited by:  Papers (8)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1144 KB)  

    The use of adaptive decision feedback techniques in the joint estimation of data, carrier phase, and symbol timing is investigated for a 49-QPRS 90 Mbit/s digital radio. The performance of the carrier and timing recovery loops are analyzed, first in a multipathfree environment and then for the case of a multipath distorted signal and a fixed decision feedback equalizer. Finally, the joint problem of data recovery, estimation of carrier phase, and symbol timing for a View full abstract»

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  • Data Compression Using Adaptive Coding and Partial String Matching

    Publication Year: 1984 , Page(s): 396 - 402
    Cited by:  Papers (200)  |  Patents (36)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (896 KB)  

    The recently developed technique of arithmetic coding, in conjunction with a Markov model of the source, is a powerful method of data compression in situations where a linear treatment is inappropriate. Adaptive coding allows the model to be constructed dynamically by both encoder and decoder during the course of the transmission, and has been shown to incur a smaller coding overhead than explicit transmission of the model's statistics. But there is a basic conflict between the desire to use high-order Markov models and the need to have them formed quickly as the initial part of the message is sent. This paper describes how the conflict can be resolved with partial string matching, and reports experimental results which show that mixed-case English text can be coded in as little as 2.2 bits/ character with no prior knowledge of the source. View full abstract»

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  • An Improved Adaptive Predictor in DPCM Based on the Kalman Filter and Its Application to Handwriting Signal Encoding

    Publication Year: 1984 , Page(s): 484 - 488
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (720 KB)  

    A method of handwriting signal encoding based on adaptive linear predictive coding (ALPC) is studied. The ALPC is a form of DPCM which uses a sequentially adaptive predictor in which a sequential estimation algorithm is used to update predictor coefficients. To improve the estimates of the predictor coefficients in the presence of quantization noise, Kalman filtering has been investigated for its feasibility. This results in improvements of not only the estimation of the predictor coefficients, but the signal-to-quantization-noise ratio (SNR) of the signals reconstructed at the receiver as well. Computer simulations have verified that the ALPC system employing the Kalman filter promises high performance and feasibility at the rate of 192 bits/s when applied to handwriting signal encoding. View full abstract»

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  • A Low Complexity Stack Decoder for a Class of Binary Rate (n-1)/n Convolutional Codes

    Publication Year: 1984 , Page(s): 476 - 479
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (432 KB)  

    In this paper we discuss the implementation of a modified stack decoder for a class of binary rate R = (n - 1)/n convolutional codes used on a binary symmetric channel (BSC). For large values of n , the classical implementation of the stack decoder quickly becomes impractical, as each extension of an information sequence estimate gives rise to 2^{(n-1)} successor estimates. A Fano type of sequential decoder is then preferable. However, by using the structure of a class of systematic rate (n - 1)/n codes, with optimum distance profile (ODP), we are able to modify the classical stack decoder such that it is of comparable complexity. The average number Of stack reorganizations, as well as the average number of successors per extension, can be reduced considerably, without increase of decoding error probability. View full abstract»

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  • Synchronization and Specification Issues in Protocol Testing

    Publication Year: 1984 , Page(s): 389 - 395
    Cited by:  Papers (33)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (856 KB)  

    Protocol testing for the purpose of certifying the implementation's adherence to the protocol specification can be done with a test architecture consisting of remote tester and local responder processes generating specific input stimuli, called test sequences, and observing the output produced by the implementation under test. It is possible to adapt test sequence generation techniques for finite state machines, such as transition tour, characterization, and checking sequence methods, to generate test sequences for protocols specified as incomplete finite state machines. For certain test sequences, the tester or responder processes are forced to consider the timing of an interaction in which they have not taken part; these test sequences are called nonsynchronizable. The three test sequence generation algorithms are modified to obtain synchronizable test sequences. The checking of a given protocol for intrinsic synchronization problems is also discussed. Complexities of synchronizable test sequence generation algorithms are given and complete testing of a protocol is shown to be infeasible. To extend the applicability of the characterization and checking sequences, different methods are proposed to enhance the protocol specifications: special test input interactions are defined and a methodology is developed to complete the protocol specifications. View full abstract»

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  • A Simple Approximation for Minimum Mean-Square Error Symmetric Uniform Quantization

    Publication Year: 1984 , Page(s): 470 - 474
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (432 KB)  

    In the context of minimum mean-square error symmetric uniform quantization, we show that for several different distributions on the input signals, log-log plots of step size versus number of output levels and mean-square error versus number of output levels both exhibit nearly linear behavior. This observation results in a straightforward design procedure for symmetric uniform quantization. View full abstract»

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  • Intermodulation Interference-Minimum Frequency Assignment for Satellite SCPC Systems

    Publication Year: 1984 , Page(s): 462 - 468
    Cited by:  Papers (13)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (744 KB)  

    This paper addresses frequency assignment for singlelevel and multilevel SCPC systems. In order to obtain the optimum channel allocation, where the influence of intermodulation noise on carriers is minimized, a practical and effective method is proposed. The quasi-optimum, solutions obtained with the proposed method are presented for single-level SCPC systems, showing their advantage in intermodulation noise reduction. Concerning frequency assignment for multilevel SCPC systems, two strategies to realize the-quasi-optimum channel allocation are compared with regard to the improvement in carrier-to-intermodulation noise power ratio. The performance of the obtained channel allocation indicates the effectiveness of the proposed optimization method. View full abstract»

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  • DPCM Picture Coding with Two-Dimensional Control of Adaptive Quantization

    Publication Year: 1984 , Page(s): 457 - 462
    Cited by:  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (816 KB)  

    With DPCM, it is advantageous to adapt the quantizer characteristic to the signal. We have studied several schemes where a gain constant is controlled by either forward or backward estimation. With forward estimation, the gain is calculated on a two-dimensional block of data, and we have studied how the form and size of the block affects performance. With backward estimation, the Jayant algorithm has been extended from one to two dimensions by letting the change in gain constant depend upon prediction errors from neighboring elements on the present and previous line. Simulations are performed on a sequence of pictures from a monochrome videophone scene with very active movement. The improvement in mean square error with adaptive control is between 2 and 5 dB compared to the performance with a uniform fixed quantizer. The subjective improvement is substantial. View full abstract»

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  • On Differential Detection of Binary FM

    Publication Year: 1984 , Page(s): 469 - 470
    Cited by:  Papers (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (232 KB)  

    Differential detection of binary frequency-shift keyed signals (binary FM) is analyzed. It is shown that, by proper selection of the delay interval, the error rate of differentially detected binary FM can be made equal to the error rate of binary DPSK provided that the modulation index h satisfies h \geq 0.5 . If h > 0.5 , it is shown that the differential detection of binary FM yields poorer performance than DPSK. View full abstract»

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  • Image Decimation and Interpolation Techniques Based on Frequency Domain Analysis

    Publication Year: 1984 , Page(s): 479 - 484
    Cited by:  Papers (9)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (880 KB)  

    A scheme for a spatial domain image data preprocessing decimation and postprocessing interpolation is presented. The scheme is implemented by appropriate FIR digital filters. Frequency and patial domain specifications are discussed in the design of the corresponding digital filters. Fast approximation techniques in the spatial domain are presented. View full abstract»

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  • A Multisubscriber Variable-Rate Sampling HCDM System with Dynamic Buffer Control

    Publication Year: 1984 , Page(s): 403 - 410
    Cited by:  Papers (1)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (896 KB)  

    In this paper we present a multisubscriber variable-rate sampling hybrid companding delta modulation (HCDM) system for simultaneous transmission of several speech signals. This system employs both the statistical multiplexing and variable-rate sampling schemes. It transmits speech signals synchronously at a fixed rate using a buffer. In this system the sampling rate of each subscriber is varied according to the speech activity and the status of buffer occupancy, and only the speech portion is coded for transmission. To optimize the system performance within the allowed maximum transmission delay (300 ms), an efficient dynamic buffer control algorithm is proposed. When the number of subscribers is six and the transmission rate for each subscriber is 16 kbits/s, the proposed system yields a performance improvement of about 10 dB over the conventional single-subscriber HCDM system. The buffer delay in this case is 150 ms, which gives a perceptually negligible effect. View full abstract»

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  • Outage Probability in Mobile Telephony Due to Multiple Log-Normal Interferers

    Publication Year: 1984 , Page(s): 380 - 388
    Cited by:  Papers (85)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (840 KB)  

    The mobile radio channel is characterized by three important factors: path losses larger than free space, fading typically taken as Rayleigh, and shadowing generally characterized as lognormal. For cellular systems, in order to determine acceptable reuse distances between base stations and to compare modulation methods, the probability of unacceptable cochannel interference (outage probability) has to be determined in the realistic situation where both fading and shadowing occur. In this paper, the average outage probability is computed for centrally located base stations when multiple log-normal interferers are present. This is done for both the mobile-to-base and base-to-mobile communication links. An unexpected result of this study is that the outage probabilities for the two cases do not differ in a significant way. Cumulative probability curves of the short-term average-signal-toaverage-interference ratio (SIR) are presented for a variety of system parameters: channel set number, propagation law exponent (γ), and dB spread (σ) of the log-normal distribution for the signal and interferers. An important observation is the large sensitivity of the performance curves to the propagation parameters: for a system with seven channel sets with a 10 dB SIR threshold, the average outage probability varies from 10 percent for \gamma = 3.7, \sigma = 6 dB, to 70 percent for \gamma = 3, \sigma = 14 dB. Alternatively, for a fixed outage objective of 10 percent, the required SIR threshold value ranges from -17 dB to 11 dB, depending on the propagation parameters. These variations make it imperative that accurate measurements of these parameters be obtained for the different service areas. Outage probabilities are also easily related to specific modulation methods and diversity approaches; detailed results are given for several representative cases. View full abstract»

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  • Constant Envelope Signal Generator with Improved Phase Linearity

    Publication Year: 1984 , Page(s): 488 - 490
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (288 KB)  

    In a recent paper entitled "Generation of constant envelope signals," Dickerson and Warren introduced a technique for the generation of a constant envelope signal which does not require hard limiting. The nonlinear distortion of the resultant phase modulator is, however, quite high. In this correspondence, we suggest two simple but effective modifications to improve the phase linearity of the modulator. View full abstract»

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  • A Power Series Expansion Approach to Evaluate Modulation Transfer Noise Effects from Digital Carriers to FDM/FM Carriers in Memoryless Nonlinear Devices

    Publication Year: 1984 , Page(s): 346 - 353
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (632 KB)  

    This paper presents a power series expansion approach for evaluating modulation transfer noise effects from L digital carriers (continuous or bursty) to K FDM/FM carriers in a common memoryless nonlinear amplifier. The method allows for a complete characterization and assessment of this type of transmission impairment and its effects in satellite communications systems. The baseband modulation transfer noise in the FM carriers caused by the random envelope fluctuations due to filtering of PSK pulses and/or by the on-off bursting of the digital carriers can be calculated. Illustrative numerical results indicate that the mixed PSK-FM mode of operation in satellite transponders should be employed with great care since a number of 4 kHz channels in the basebands of the FM carriers may receive prohibitive interference. View full abstract»

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  • Self-Orthogonalizing Adaptive Equalization in the Discrete Frequency Domain

    Publication Year: 1984 , Page(s): 371 - 379
    Cited by:  Papers (17)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1008 KB)  

    A self-orthogonalizing discrete adaptive equalizer for synchronous data transmission is presented, based on the overlapsave filtering technique. Self-orthogonalization in the discrete frequency domain is adaptively performed by premultiplying by a diagonal matrix the MSE gradient estimates before projecting by means of Rosen's gradient projection method. The diagonal of the matrix is the inverse of the power spectrum of the received sequence taken at equally spaced frequencies, and estimates are obtained by using Bartlett's procedure of periodograms averaging for spectrum estimation. Projection is accomplished by means of an off-line derived projection matrix. Confidence of gradient estimates is improved by means of a block correlated estimation technique using available DFT's of blocks of data. This equalizer is compared to time-domain self-orthogonalizing algorithms as regards speed of convergence and ease of implementation. During the short startup phase, convergence is competitive with that of Godard's algorithm, which is the fastest algorithm known, and in the decision-directed mode, fast convolution performed by blocks results in a considerable reduction of the number of multiplications with respect to the time domain algorithms. A computationally simpler, unconstrained startup algorithm is also examined, obtained by removing the projection while retaining gradient block estimates in the DFD. View full abstract»

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Aims & Scope

IEEE Transactions on Communications focuses on all telecommunications including telephone, telegraphy, facsimile, and point-to-point television by electromagnetic propagation.

 

 

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Editor-in-Chief
Robert Schober
University of British Columbia