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IEE Proceedings I (Communications, Speech and Vision)

Volume 138, Issue 6, December 1991

Volumes & issues:

Volume 138, Issue 6

December 1991

Analysis of asynchronous multiplexers for periodic traffic in integrated broadband networks
Queuing analysis of the selective repeat ARQ scheme
Efficient flow control in a multiclass telecommunications environment
Simulation study of trellis-coded transceivers for high-bit rate digital subscriber loops
Throughput efficiency of a class of continuous ARQ schemes under markovian error patterns
  • Author(s): Tsern-Huei Lee
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 511 –516
  • DOI:  10.1049/ip-i-2.1991.0068
  • Type: Article
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  • p. 511 –516 (6)
    The independent error model is often adopted to simplify the analysis of ARQ strategies. This assumption, however, is not true for channels where transmission errors occur in bursts. The limiting throughput efficiency of a class of continuous ARQ schemes with repeated transmissions is analysed under Markovian error patterns. The scheme without repeated transmissions, i.e. the classic go-back-N scheme, has been analysed in a previous paper. The results show that the throughput efficiency can be significantly increased by transmitting multiple copies of each data block contiguously to the receiver. The results obtained are also compared under Markovian error patterns with those obtained under the independent error model.
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Correlation decoding of the (24, 12) Golay code using neural networks
Channel-estimation-based adaptive DFE for fading multipath radio channels
  • Author(s): P.K. Shukla  and  L.F. Turner
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 525 –543
  • DOI:  10.1049/ip-i-2.1991.0070
  • Type: Article
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  • p. 525 –543 (19)
    The performance of an adaptive decision feedback equaliser (DFE) over an HF channel when its taps are derived from an estimate of the sampled channel impulse response is examined. The conventional approach to adaptation is to adjust the equaliser taps directly to minimise a least squares error cost function. It is shown that the channel estimate approach not only yields superior performance, but also involves fewer computations than the conventional square-root-Kalman approach. The problem of channel estimation is mathematically analysed for both the steepest descent and recursive least squares algorithms, and theoretical results are compared with simulation. It is also shown how simple theoretical predictions of the performance of the DFE can be made when using this alternative method of implementation.
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Signalling efficiency modelling and measurement of simplex block encoded MBC systems
  • Author(s): P.S. Cannon  and  A.H. Dickson
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 544 –548
  • DOI:  10.1049/ip-i-2.1991.0071
  • Type: Article
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  • p. 544 –548 (5)
    A theoretical technique is developed to estimate the signalling efficiency of a meteor burst communications (MBC) system which uses a block encoded simplex protocol over underdense trails. The signalling efficiency is determined for various information block durations and compared to experimental results from the BLOSSOM (beyond line of sight signalling over meteors) MBC system, which could vary its information block duration. Although the general shape of both the theoretical and experimental results are similar there are detailed differences. Reasons for these discrepancies are suggested.
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Characterisation of mobile radio signals: model description
  • Author(s): J.D. Parsons  and  A.M.D. Turkmani
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 549 –556
  • DOI:  10.1049/ip-i-2.1991.0072
  • Type: Article
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  • p. 549 –556 (8)
    Following a brief review of multipath propagation models, the major characteristics of mobile radio signals are determined using an adaptation of the generic three-dimensional model proposed by Aulin. The various multipath component waves, which have equal amplitude and random, uniformly-distributed phase are allowed to have spatial angles of arrival with both vertical and horizontal components. The spatial angle of arrival in the horizontal plane is assumed to be uniformly distributed in the interval (0, 2π) but a new PDF is proposed for the vertical angle of arrival, β. The effective scatterers are assumed to form a cyclinder around the mobile with a radius and height that depends on the maximum value βm of the vertical angle of arrival. Suitable values for βm are estimated from published experimental results. The limitations of two-dimensional scattering models are illustrated by considering not only the power spectral density, but also the cross correlation between antennas spatially separated in the vertical plane.
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Characterisation of mobile radio signals: base station crosscorrelation
  • Author(s): A.M.D. Turkmani  and  J.D. Parsons
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 557 –565
  • DOI:  10.1049/ip-i-2.1991.0073
  • Type: Article
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  • p. 557 –565 (9)
    A three-dimensional model of mobile radio propagation has been used to study the crosscorrelation between signals received on two spatially separated antennas at a base station site. The factors that have the greatest influence have been identified. It has been shown that, although vertical separation can be used to obtain signals that are sufficiently decorrelated to produce significant diversity gain, horizontal separation is still generally preferable. A vertical separation of 20 wavelengths (20 λ) results in a crosscorrelation which is always less than 0.7, irrespective of mobile location, in a cell of radius 1 km, but the same value is only obtainable from 50% of locations if the cell radius is doubled. For horizontally separated antennas, the correlation coefficient is less than 0.7 for signals received from 98% of locations in a cell of radius 1 km, and this is only reduced to 87% of locations if the radius is increased to 5 km. In some circumstances, composite separation (vertical plus horizontal) may be useful.
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Equaliser techniques for QAM transmissions over dispersive mobile radio channels
  • Author(s): W.T. Webb  and  R. Steele
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 566 –576
  • DOI:  10.1049/ip-i-2.1991.0074
  • Type: Article
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  • p. 566 –576 (11)
    Channel equalisation of 32 Mbit/s data transmissed via 16-level QAM at 1.9 GHz over dispersive fading channels is addressed using computer simulations. The multipath propagation channels used had three independent Rayleigh fading paths. The conventional equaliser systems were found to be unsatisfactory and an alternative system was conceived based on a RAKE-type equaliser controlling an iterative decision feedback equaliser. Although complex, simulations showed this equaliser system to yield an order of magnitude reduction in BER compared to that for a linear equaliser at SNRs in excess of 15 dB. Advantages are presented of using error correction coding in the conventional way and in an iterative manner to aid convergence of the equaliser. Gains of 2 to 3 dB were achieved by iterative decoding but at the expense of considerable increase in complexity. When second order spatial diversity was used we achieved improvements of about 5 to 7 dB when both systems were uncoded despite the inherent diversity that can be gained from equalising a wideband channel. By operating at SNRs greater than 25 dB, we were able to simulate transmission at 32 Mbit/s with a BER below 10−3.
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Radio transmission at 1800 MHz into, and within, multistory buildings
  • Author(s): A.M.D. Turkmani  and  A.F. de Toledo
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 577 –584
  • DOI:  10.1049/ip-i-2.1991.0075
  • Type: Article
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  • p. 577 –584 (8)
    Investigations of propagation into, and within, buildings at 1800 MHz have been undertaken using buildings in the University of Liverpool precinct. Measurements of the mean signal level have been made in rooms and corridors of four different buildings and, where appropriate, these have been compared with measurements at street level outside. The composite Rayleigh-pluslog-normal distribution successfully modelled the measured cumulative distributions of all measurements. The average measured penetration loss at ground floor level was found to be 13 dB. The rate of change of penetration loss with height was −1.4 dB per floor. For floor levels higher than the sixth floor, it was found to be 0.4 dB per floor. The rate of change of the mean signal level for signals travelling within buildings was, on average, 8.3 dB per floor. The best coverage was obtained by locating the transmitter in a large room at the centre of the building. The path loss attenuation factor n that best modelled the within-building measurements was found to be 5.6.
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Subjective quality quantification of digital voice communication systems
  • Author(s): Spiros Dimolitsas
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 585 –595
  • DOI:  10.1049/ip-i-2.1991.0076
  • Type: Article
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  • p. 585 –595 (11)
    The expanding digitisation of the telephone networks has brought about an increasing interest in methods addressing the performance characterisation of new digital voice transmission systems. In the paper a review of subjective methodologies for the assessment of telephone quality digital voice communications systems is provided. Technical aspects concerning the network applications and other characteristics relevant to the type of system under evaluation are considered first, since these factors influence the selection of a suitable assessment methodology. Next, listener opinion and conversational opinion tests are described, with particular emphasis on their respective merits, characteristics and differences. Finally, articulation and diagnostic tests, while not generally employed for the assessment of telephone quality, are briefly examined, since the results obtained from such tests are often correlated with the perceptual performance of telephone quality communications links.
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Algorithm for high precision root finding from high order LPC models
Finite state CELP for variable rate speech coding
  • Author(s): S.V. Vaseghi
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 603 –610
  • DOI:  10.1049/ip-i-2.1991.0078
  • Type: Article
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  • p. 603 –610 (8)
    The performance of a variable rate code excited linear predictor system is investigated. The coding system is based on a finite state CELP (FSCELP) frame work. Each individual state is primarily identified with a LPC model order, LPC coefficients bit allocation, excitation code book population density and state encoding rate. Successive input speech vectors are encoded at a rate that depends on the current state of the FSCELP system and the input vector characteristics. The use of a finite state system involves implicit clustering of speech signals. The lower rate states are selected during highly correlated steady state speech segments when relatively few bits are required to obtain adequate fidelity. For speech signals with a strong glottal excitation, unvoiced signals and transient speech segments, a relatively greater quantisation accuracy is needed to obtain good fidelity and therefore higher rate states of the system are used. Further improvement is obtained by using gamma populated excitation codebooks, for those states that are mainly used to encode speech signals with a strong underlying glottal excitation pulses. Experiments focus on investigation of the varying encoding requirements of the excitation signal for low pass, voiced, unvoiced and transient speech signals. The parameters of the finite state CELP system are designed to match the encoding requirements of typical speech signals. The greater part of the coding gain is obtained from variable rate encoding of the excitation signal. Using a six-state FSCELP, good quality speech is obtained at an average, maximum and minimum bit rates of 4 kbit/s, 10 kbit/s and 2 kbit/s, respectively.
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Two-stage convolver-based DCT and IDCT algorithms
  • Author(s): J.-L. Wu ; S.-H. Hsu ; W.-J. Duh
  • Source: IEE Proceedings I (Communications, Speech and Vision), Volume 138, Issue 6, p. 611 –614
  • DOI:  10.1049/ip-i-2.1991.0079
  • Type: Article
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  • p. 611 –614 (4)
    Discrete cosine transform is the most popular solution for image coding owing to its near optimal performance, yet it is not signal dependent. A two-stage convolver-based DCT and IDCT algorithm with power-of-two length is presented. The transform matrix of IDCT is decomposed into the product of pre- and post processing matrices by using Möbius inversion formula. The pre-processing stage consists of just additions and subtractions; the post-processing stage performs circular convolution operations. A nearly recursive computational method for preprocessing calculation is given which considerably reduces the operation count and is valuable in practical usage. The post-processing is a circular convolution and can be computed using number theoretic transform, (semi-) systolic arrays or transversal filters.
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Parallel implementation of a fast thinning algorithm using image compression

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