Communications, Speech and Vision, IEE Proceedings I (IEE Proc Comm Speech Vis)
Description
Discontinued in 1994. Continued by IEE Proceedings - Communications (1350-2425). Classification: Speech processing systems; Telecommunication systems.
- WebsiteIEE Proceedings - Part I: Communications, Speech & Vision website
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Other titlesIEE proceedings. I, Communications, speech, and vision, Institution of Electrical Engineers., Communications, speech, and vision
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ISSN0956-3776
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OCLC19239750
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Material typePeriodical, Internet resource
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Document typeJournal / Magazine / Newspaper, Internet Resource
Publications in this journal
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Article: New LSP encoding method based on two-dimensional linear prediction
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ABSTRACT: A new line spectrum pair (LSP) encoding method, two-dimensional differential LSP (2DdLSP) coding, is proposed. This 2DdLSP approach simultaneously takes advantage of strong interframe and intraframe correlation of LSP parameters. The parameters to be quantised are the prediction residuals which possess the property of smaller variance than LSP parameters. This is the key point for the success of the 2DdLSP method. Three quantisation schemes, a scalar quantisation and two vector quantisations (VQs), are presented. The best result in simulation is achieved by using the product codebook VQ. A spectral distortion of 1 dB<sup> 2</sup> at 19 bits per frame can be obtained when the frame period is 10 ms. Both within-training and out-of-training tests have demonstrated the robustness of the proposed method to data variation.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: HiMA: a hierarchical and modular ATM switch with partially shared output buffer
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ABSTRACT: The authors propose a hierarchical and modular ATM switch for BISDN. To improve the queueing performance, they adopt the output queueing technique and allow several outputs to partially share the same output buffer space. The performance of the proposed switch is evaluated under uniform and nonuniform traffic patterns. Comparisons with the knockout switch, recursive switch, SCOQ, and Christmas-tree switch show that, in terms of complexity, crosspoint count, scalability and synchronisation, the proposed switch is superior.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Two-level decoding of (32, 16, 8) quadratic residue code
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ABSTRACT: The Pless (1986) construction of the (24, 12, 8) extended Golay code can be modified to construct the (32, 16, 8) extended QR code. Some structural similarities are discovered between the (32, 16, 8) QR code and the (24, 12, 8) Golay code. A simple decoder is developed based on the two-level structure of the (32, 16, 8) QR code. In addition to correcting all three or less bit errors, the proposed algorithm is able to correct some four-bit error patterns. Recently, soft-logic decoding of block codes has found increasing practical applications. The two-level structure renders an extension of Vardy and Be'ery's fast algorithm to the (32, 16, 8) QR code. The Vardy-Be'ery's (1991) algorithm is, so far, the most efficient scheme for the Golay code in the literature, with direct application impact. Its extension to the (32, 16, 8) code is also computationally efficient, as analysed in detail.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Multilayer perceptron based decision feedback equalisers for channels with intersymbol interference
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ABSTRACT: The paper describes the application of multilayer perceptrons to the problem of adaptive channel equalisation in digital communications systems. In particular, the use of decision feedback structures is investigated for time-invariant and time-variant bandlimited channels. Simulation results show that the equaliser based on the multilayer perceptron provides better bit error rate performance compared with the conventional decision feedback equaliser if the equaliser length corresponds to the time spread of the channel. Increasing the equaliser length leads to equivalent bit error rates for the decision feedback and the multilayer perceptron based equaliser. Because of the more complex structure of the multilayer perceptron its adaptive behaviour for time-variant channels is inferior compared with the decision feedback equaliser. Nonlinearities on the transmission channel can be equalised much better by the multilayer perceptron based equaliser.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Flow control methodology for virtual-circuit networks
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ABSTRACT: An input rate flow control methodology for a virtual-circuit computer network is presented. Power is used as the performance measure of a virtual circuit and network power, which is the product of all of the virtual circuit powers, is used as the network performance measure. It is shown that the network power has a unique maximum point in the feasible region of the throughput space. A distributed flow control algorithm is presented and proof of convergence is given. Several numerical examples illustrate the algorithm.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Modelling of radio transmissions into and within multistorey buildings at 900, 1800 and 2300 MHz
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ABSTRACT: A multiple regression analysis method is used to predict the signal path loss encountered by radio transmissions into and within buildings at 900, 1800 and 2300 MHz. It has been found that in addition to the initial free space path loss, the floor mean signal level for radio transmissions into buildings is also dependent on the logarithm of the floor area and on the condition of transmission, which is represented in this study by the number of building sides that have a direct line of sight to the transmitter. The path loss for radio transmissions within buildings has been found, however, to be dependent on the logarithm of the distance, the floor area, the number of floors between the transmitter and the receiver, and on two other factors identified in the study as sight and ground. The first factor accounts for signals leaving and returning to the building, complemented by some considerations on the ability of signals to propagate on the floor where the transmitter was located. The ground factor accounts for the observed tendency for the signal strength to be higher on the first two floors of a building. Statistical methods have also been used to determine the reliability of the prediction.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Trans: a reliable broadcast protocol
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ABSTRACT: Trans is a novel reliable broadcast protocol for the data-link layer of the ISO protocol hierarchy. The protocol exploits the broadcast nature of the physical communication media typically used in local-area networks. A combination of positive and negative acknowledgment strategies with transitivity of positive acknowledgments provides reliable operation without requiring a separate acknowledgment from every recipient of a message frame. Trans achieves reliable communication despite a noisy communication medium and processor fail-stop, omission and timing faults. Multicast, rather than fully broadcast, communication is readily obtained by operating several subnetworks over the same local-area network or, alternatively, by including a destination address list in the header of each broadcast frame.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Packet reservation protocols for multichannel satellite networks
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ABSTRACT: Owing to system economics, many satellite data networks have star configurations, with low cost user terminals communicating with high performance hub earth stations over inbound multiaccess channels and outbound broadcast channels. The inbound channel data rate is often limited by the power budget, so that multiple satellite channels are required to accommodate the overall traffic in a large network. Most packet multiaccess protocols previously considered, when applied to multichannel satellite networks, enable distinct groups of users to share individual channels. In contrast, this paper presents a new class of multichannel reservation protocols (MRPs) which improves channel utilisation by enabling complete sharing of all channels among all users. For good throughput efficiency and reduced delay, the MRPs employ slotted random access with immediate retransmissions for first-packet and reservation request transmissions, and reservation of consecutive time slots for transmissions of remaining packets in multipacket messages. Delay throughput characteristics are derived under steady state conditions, Numerical results show that the MRPs achieve higher throughput capacity than random access protocols and lower average delay than other reservation protocols.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Multiple-capture performance of DS-SS packet radio systems with common spreading code
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ABSTRACT: The author presents a multiple-capture model for the centralised spread-spectrum (SS) packet radio systems in which each user is assigned common direct-sequence (DS) spreading code for packet transmission. Basic equations for the multiple-capture probability and throughput performance are derived at the central receiver. At the link-level, two key design parameters in common spreading code systems are evaluated, namely, the expected number of packets captured at the receiver and the allowable number of simultaneous transmissions that are supportable at a specified data bit-error rate and probability of packet capture. Using the block oriented systems simulator (BOSS), simulations were carried out for the central receivers with coherent, differential, and envelope header detection, and results from the simulations are compared with theoretical evaluations of the multiple-capture probability. It is shown, for a threshold approximation to the probability of data packet success, that the multiple-capture model significantly improves system throughput.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Sample rejection and importance sampling in the simulation of multidimensional signalling systems
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ABSTRACT: Sample rejection has been proposed as a means for improving the efficiency of computer simulation used for error rate estimation. Previous work has not examined quantitatively the feasibility of sample rejection for improving the efficiency of simulation of multidimensional signalling schemes. The present work aims to determine the usefulness of sample rejection for the simulation of such systems. Two methods based on sample rejection are described for the Monte Carlo simulation of small error probabilities, P<sub> e</sub> in digital communication systems. The first is based on the observation that when P<sub> e</sub> is small most of the noise vectors are known in advance not to cause errors, and consequently need not be simulated. Discarding such noise vectors will result in savings in computer simulation time. The second method is based on generating noise vectors whose probability density function has a hole carved around the origin. This method replaces the original noise input density function by a biased noise input density and is a form of importance sampling. The expected savings in computer time achieved by use of these methods is investigated. Quantitative results are obtained for multidimensional signalling schemes without memory. The expected savings are compared to those achieved by the use of conventional importance sampling. The suitability of the two methods for simulation of multidimensional systems with memory is considered.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Codeword synchronisation of CCMA scheme
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ABSTRACT: A kind of codeword synchronisation scheme in decoding adder channel collaborative coding multiple access (CCMA) sequence is presented. By making use of the correlation between adjacent added codewords and the shifting properties of the received sequence, the scheme can provide very good codeword synchronisation information about the received data stream. Simulated results in various noisy environments show that the scheme is very effective.< >Communications, Speech and Vision, IEE Proceedings I 01/1994; -
Article: Efficient fast packet switch fabric with shared input buffers
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ABSTRACT: The authors deal with an efficient high-speed packet switching fabric suitable for applications in future high-speed networks. An advanced implementation architecture based on shared buffers at each input is studied. An important result is that the proposed switching fabric achieves optimum throughput-mean switching delay performance jointly with reduced buffering requirements and without having to resort to a faster switching fabric.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Multichannel adaptive forward error-correction system
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ABSTRACT: A new multichannel adaptive error-correction system has been investigated for use over noisy time-varying channels. Information is fed at a fixed overall rate over a number of parallel channels, with the amount of information per channel being determined on the basis of channel quality. The system is shown to be superior to that obtainable with fixed-rate forward error-correction systems, and in addition the proposed system is shown to have a performance superior to that obtainable with classical diversity systems operating at the same rate.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Multipath fading effects on carrier recovery of BPSK signal in digital radio
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ABSTRACT: Tracking performance of the I-Q carrier recovery loop for BPSK signal in band-limited channel with multipath interference is analysed. The multipath fading channel is characterised by the Rummler's simplified three-ray model (1979) for line-of-sight digital radio system. A closed-form expression for the phase-error variance related to the multipath interference, intersymbol interference and channel noise is derived. Excess phase offset induced from the multipath interference is also examined. The closed-form expression of the jitter variance can be evaluated numerically to assess the degradation of synchroniser performance. Asymptotic limits of the performance are also discussed.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Trellis in-loop PARCOR coefficients quantiser for speech signals
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ABSTRACT: A newly developed PARCOR coefficients quantisation scheme is presented which is referred to as trellis in-loop PARCOR coefficients quantisation. In the in-loop quantisation scheme, each PARCOR coefficient is determined and quantised inside the analysis computation loop prior to the high order one. Therefore, the quantisation error is taken into account at each stage of lattice filter analysis. This scheme, when combined with trellis structure and associated with appropriate search algorithms, can achieve lower distortions than the conventional scalar quantisation. Simulation results show that by incorporating trellis structure into the in-loop quantisation scheme and using (M, L)-search algorithm with M=4 and L=8, an average gain of six bits per frame over the traditional methods can be achieved.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Efficient simulation of multilevel and coded transmission systems
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ABSTRACT: The authors discuss how an asymptotically efficient simulation technique, originally proposed for binary communication systems, can be applied also to modulation systems with a large number of signalling waveforms, retaining much of the efficiency of the method. Basically, the probability of error is decomposed into sums of terms corresponding to all possible errors. Then each term is evaluated by a series of ad hoc experiments, designed as if the modulation system were binary. For a given computational effort, each experiment is performed an appropriate number of times to minimise the variance of the estimated probability of error. When the number of waveforms is very large, they show that random selection of the experiments can provide an unbiased form of stochastic truncation of the sum. The authors present examples of application to block codes, convolutional codes, and trellis coded modulation. These can be easily handled, provided sufficient information about the code structure is available.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Blocking, hand-off and traffic performance for cellular communication systems with mixed platforms
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ABSTRACT: Cellular communication systems that support a mixture of platform types distinguished by different mobility characteristics are considered. A tractable analytical model for traffic performance analysis is developed using multidimensional birth-death processes and the method of phases. The framework allows consideration of homogeneous and nonhomogeneous systems, a broad class of dwell time distributions, and 'missed' hand-off initiations. Cut-off priority for hand-offs and several platform types are considered to demonstrate the approach. The effects of different mobility parameters and of imperfect detection of hand-off needs are examined. Theoretical performance characteristics are obtained. These exhibit carried traffic, hand-off activity, blocking probability and forced termination probability for each platform type. The realisable exchange of blocking for hand-off performance is shown. Computational issues are discussed and a method for extrapolating performance characteristics for systems with many channels is presented.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Speech modelling using cepstral-time feature matrices in hidden Markov models
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ABSTRACT: The paper explores the use of 2-dimensional cepstral-time features for the utilisation of correlation among successive speech spectral vectors, within a hidden-Markov-model (HMM) framework. A cepstral-time-feature matrix is obtained from a 2-dimensional discrete cosine transform of a spectral-time matrix. Advantages of cepstral-time features are that cepstral-time-feature matrices are a simple and robust method of representing short-time variation of speech spectral parameters; a cepstral-time matrix contains information on the transitional dynamics of feature vectors within the matrix; speech recognition based on cepstral time matrices is more robust in noisy environments; and use of a matrix of M cepstral vectors implies a minimum HMM-state duration constraint of M vector units. A simple framework investigated in the paper for applications of cepstral-time features is a finite-state-matrix quantiser (FSMQ), a special case of the HMM. It is used for initialisation of the training phase of HMMs.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Optimal convolution filters and an algorithm for the detection of wide linear features
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ABSTRACT: The author presents the theory for developing optimal convolution filters for the identification of wide linear features and an algorithm for distinguishing them from other types of image feature. Some filters are given and the results of applying the algorithm to some real images are shown. These filters, and the algorithm, are appropriate for use in knowledge based systems which aim at identifying roads, canals, hedges and rivers in images, or in digitised maps.< >Communications, Speech and Vision, IEE Proceedings I 11/1993; -
Article: Type-II hybrid ARQ protocol using concatenated codes
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ABSTRACT: The high throughput achieved by type-II hybrid automatic repeat request (ARQ) protocols can be enhanced by using the inner structure of a codeword in a concatenated coding scheme. The paper presents a new type-II hybrid scheme based on concatenated codes to which two retransmission strategies have been applied. The results on throughput bounds show that the proposed protocols are effective even in the case of communication channels with high error rates.< >Communications, Speech and Vision, IEE Proceedings I 11/1993;
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