Showing papers in "IEEE Journal on Selected Areas in Communications in 1992"

A PRML system for digital magnetic recording

Abstract: The realization of a digital recording system using partial-response class-IV signaling with maximum-likelihood sequence detection (MLSD) is described. To perform MLSD at the high data rates encountered in recording systems, a simple implementation of the Viterbi detector based on a difference-metric algorithm is developed. Decision-directed schemes for gain control and timing recovery, for tracking variations of the gain and timing phase during data readback, and for fast initial adjustment from a known preamble are presented. The dynamic behavior of the control algorithms was studied by computer simulations. Coding was used to facilitate timing recovery and gain control, to limit the path memory length of the Viterbi detector, and to allow fast and reliable startup of the receiver. The design and properties of rate 8/9 constrained codes are examined. The problem of equalization is addressed, and analog and combined analog/digital filter implementations are developed. A simple adaptive equalizer capable of compensating variations of the recording channel characteristics with track radius and/or head-to-medium distance is described. >

An objective measure for predicting subjective quality of speech coders

Abstract: A perceptually motivated objective measure for evaluating speech quality is presented. The measure, computed from the original and coded versions of an utterance, exhibits statistically a monotonic relationship with the mean opinion score, a widely used criterion for speech coder assessment. For each 10-ms segment of an utterance, a weighted spectral vector is computed via 15 critical band filters for telephone bandwidth speech. The overall distortion, called Bark spectral distortion (BSD), is the average squared Euclidean distance between spectral vectors of the original and coded utterances. The BSD takes into account auditory frequency warping, critical band integration, amplitude sensitivity variations with frequency, and subjective loudness. >

A measurement-based prioritization scheme for handovers in mobile cellular networks

Abstract: A method of improving the quality of service in mobile cellular systems based on prioritization of handover requests is presented. The objective is to improve perceived quality of cellular service by minimizing both the probability of forced termination of ongoing calls due to handover failures and the degradation in spectrum utilization. A model based on a multiple-priority nonpreemptive queuing discipline is developed. New calls are blocked if all channels are occupied. Handover requests are queued such that as soon as a channel is available, it is offered to the mobile subscriber with the measurement results closest to the minimum acceptable power level for communication. Service rate is given by channel occupancy time distribution and is assumed to be exponential. The performance of a cellular system employing the proposed handover policy is evaluated analytically and by simulation, and results are compared to those obtained when the cellular system employs nonprioritized call handling and first-in/first-out queuing discipline. This provides lower probability of forced termination and less call blocking, less reduction in traffic, and less delay. >

Finite-state modulation codes for data storage

Abstract: The authors provide a self-contained exposition of modulation code design methods based upon the state splitting algorithm. They review the necessary background on finite state transition diagrams, constrained systems, and Shannon (1948) capacity. The state splitting algorithm for constructing finite state encoders is presented and summarized in a step-by-step fashion. These encoders automatically have state-dependent decoders. It is shown that for the class of finite-type constrained systems, the encoders constructed can be made to have sliding-block decoders. The authors consider practical techniques for reducing the number of encoder states as well as the size of the sliding-block decoder window. They discuss the class of almost-finite-type systems and state the general results which yield noncatastrophic encoders. The techniques are applied to the design of several codes of interest in digital data recording. >

Equalizers for multiple input/multiple output channels and PAM systems with cyclostationary input sequences

Abstract: The author studies minimum mean square error (MMSE) linear and decision feedback (DF) equalisers for multiple input/multiple output (MIMO) communication systems with intersymbol interference (ISI) and wide-sense stationary (WSS) inputs. To derive these equalizers, one works in the D-transform domain and uses prediction theory results. Partial-response MMSE equalizers are also found. As an application, the author considers a pulse amplitude modulation (PAM) communication system with ISI and cyclostationary inputs. The MMSE linear and DF equalizers are determined by studying an equivalent MIMO system. The resulting filters are expressed in compact matrix notation and are time-invariant, whereas the corresponding single input/single output filters are periodically time-invariant. The author also considers MMSE equalizers for a wide-sense stationary process by introducing a 'random phase'. To aid in the performance evaluation of various equalizers, the author derives their mean square errors. >

On the feasibility of a CDMA overlay for personal communications networks

Abstract: Because of the continually increasing demand for mobile communications, it has been suggested that personal communication networks (PCNs) be established in the 1850-1990 MHz range. However, that band of frequencies is currently occupied by various microwave signals transmitted by users ranging from utility companies to state and local agencies. In order to allow both sets of users to occupy these frequencies as well as improve the spectral efficiency of this band, a spread-spectrum overlay is proposed, whereby a code-division multiple-access (CDMA) PCN would share the spectral band with the existing narrowband microwave traffic. The results of several field tests which have been designed to demonstrate the feasibility of an overlay of this type are discussed. >

Bandlimited quasi-synchronous CDMA: a novel satellite access technique for mobile and personal communication systems

Abstract: Recent trends in digital communications are opening commercial applications to code division multiple access (CDMA). A novel access technique based on bandlimited quasi-synchronous CDMA (BLQS-CDMA) is described, showing all the advantages of synchronizing conventional direct sequence CDMA to drastically reduce the effect of self-noise. Bandlimitation is achieved with no detection loss by means of Nyquist chip shaping, leading to a simple all-digital demodulator structure. Detection losses due to imperfect carrier frequency and chip timing synchronization are analytically derived and numerical results are checked by computer simulations. Impairments due to satellite transponder distortions are evaluated. The full digital modem structure is presented, together with possible applications to mobile and very small aperture terminal (VSAT) satellite communications. >

Subband image coding using entropy-coded quantization over noisy channels

Abstract: Under the assumption of noiseless transmission the authors develop two entropy-coded subband image coding schemes. The difference between these schemes is the procedure used for encoding the lowest frequency subband: predictive coding is used in one system and transform coding in the other. After demonstrating the unacceptable sensitivity of these schemes to transmission noise, the authors also develop a combined source/channel coding scheme in which rate-compatible convolutional codes are used to provide protection against channel noise. A packetization scheme to prevent infinite error propagation is used and an algorithm for optimal assignment of bits between the source and channel encoders of different subbands is developed. It is shown that, in the presence of channel noise, these channel-optimized schemes offer dramatic performance improvements. >

A low-delay CELP coder for the CCITT 16 kb/s speech coding standard

Abstract: A low-delay code-excited linear prediction (LD-CELP) speech coder which is expected to be standardized in 1992 as a CCITT G Series Recommendation for universal applications of speech coding at 16 kb/s is presented. The coder achieves a one-way coding delay of less than 2 ms by making both the LPC predictor and the excitation gain backward-adaptive and by using a small excitation vector size of five samples. The official CCITT laboratory tests revealed that the speech quality of this 16 kb/s LD-CELP coder is either equivalent to or better than that of the CCITT G.721 standard 32-kb/s ADPCM coder for almost all conditions tested. A description of the LD-CELP algorithm, its implementation on the DSP32C for CCITT testing, and performance results from these tests are presented. >

Signal compression: technology targets and research directions

Abstract: A description of technology targets in signal compression and a nonexhaustive account of research directions that may lead toward these targets are presented. Opportunities for integrating source coding and channel coding technologies are also pointed out. Such integration, which has hitherto been an informal exercise, will become increasingly essential as communication capabilities are stretched with capacity-limited channels such as wireless media. In parallel, as greater sophistication is sought in the integration of speech and data with broadband signals such as CD-audio and high-resolution video, there will be increased interaction of signal compression technology with the field of communication networking. >

Channel modeling and simulation in satellite mobile communication systems

Abstract: An analog model describing signal amplitude and phase variations on shadowed satellite mobile channels is proposed. A linear combination of log-normal, Rayleigh, and Rice models is used to describe signal variations over an area with constant environment attributes while an M-state Markov chain is applied to represent environment parameter variations. Channel parameters are evaluated from the experimental data and utilized to verify a simulation model. Results, presented in the form of signal waveforms, probability density functions, fade durations, and average bit and block error rates, show close agreement with measurements. >

New networks for perfect inversion and perfect reconstruction

Abstract: The authors present a new network structure that realizes perfect inversion networks (PINs) and perfect reconstruction networks (PRNs). In some applications, such as transform source coders, it is important that the cascade of the forward and the inverse transform give the identity exactly (perfect inversion), although the coefficients and the intermediate results are quantized. In subband coders, for example, the split and merge filter banks should preferably have perfect reconstruction. It is advantageous if perfect reconstruction can be accomplished even when the coefficients and the intermediate results are quantized. The proposed network has a ladderlike shape and a predescribed symmetry between the forward and inverse network or between the split and merge bank. In some parts of this ladder network almost any function is allowed. Due to the prescribed symmetry, the property of perfect inversion or perfect reconstruction is structurally assured. >

Suppression of near- and far-end crosstalk by linear pre- and post-filtering

Abstract: Full-duplex data communication over a multi-input/multi-output linear time-invariant channel is considered. The minimum mean square error (MMSE) linear equalizer is derived in the presence of both near- and far-end crosstalk and independent additive noise. The MMSE equalizer is completely specified in terms of the channel and crosstalk transfer functions by using a generalization of previous work due to Salz (1985). Conditions are given under which the equalizer can completely eliminate both near- and far-end crosstalk and intersymbol interference. The MMSE transmitter filter, subject to a transmitted power constraint, is specified when the channel and crosstalk transfer functions are bandlimited to the Nyquist frequency. Also considered is the design of MMSE transmitter and receiver filters when the data signals are arbitrary wide-sense stationary continuous or discrete-time signals, corresponding to the situation where the crosstalk is not phase-synchronous with the desired signal. >

A fast feature-based block matching algorithm using integral projections

Abstract: Block-by-block motion compensation algorithms are studied for video-conference/video-telephone television signals. A fast feature-based block matching algorithm using integral projections for the motion vector estimation is proposed. The proposed algorithm reduces the motion estimation computations by a factor of two by calculating the one-dimensional cost functions rather than the two-dimensional ones. Also, the low sensitivity of the proposed algorithm to the presence of additive noise is shown experimentally. Simulation results based on the original and noisy image sequences are presented. >

Performance evaluation for cellular CDMA

Abstract: The authors consider the performance of a cellular radio, direct-sequence code-division multiple access, (CDMA) system. The base-to-mobile link is modeled as a flat Rayleigh fading channel, with all signals transmitted from a given base station fading in unison. For the mobile-to-base link, the authors use a similar model, except that the waveforms from all users are assumed to experience independent fading. The effects of imperfect power control are shown. >

Performance of direct-sequence systems with long pseudonoise sequences

Abstract: A performance analysis of direct-sequence systems with long pseudonoise sequences is presented. The assessment of the symbol error probability in the presence of multiple-access interference is stressed, but other types of interference are also considered for completeness. Both binary and quaternary spreading waveforms are treated. The analysis, which uses a nonstandard Gaussian approximation, makes transparent the answers to a number of technical issues. Approximate error probabilities that are relatively simple computationally are derived. The approximations are accurate to within at least one decibel and, more typically, a few tenths of a decibel. >

A decision feedback equalizer with time-reversal structure

Abstract: This work describes the use of a receiver with a time-reversal structure for low-complexity decision feedback equalization of slowly fading dispersive indoor radio channels. Time-reversal is done by storing each block of received signal samples in a buffer and reversing the sequential order of the signal samples in time prior to equalization. As a result, the equivalent channel impulse response as seen by the equalizer is a time-reverse of the actual channel impulse response. Selective time-reversal operation, therefore, allows a decision feedback equalizer (DFE) with a small number of forward filter taps to perform equally well for both minimum-phase and maximum-phase channel characteristics. The author evaluates the theoretical performance bounds for such a receiver and quantifies the possible performance improvement for discrete multipath channels with Rayleigh fading statistics. Two extreme cases of DFE examples are considered: an infinite-length DFE; and a DFE with a single forward filter tap. Optimum burst and symbol timing recovery is addressed and several practical schemes are suggested. Simulation results are presented. The combined use of equalization and diversity reception is considered. >

Symbol-aided plus decision-directed reception for PSK/TCM modulation on shadowed mobile satellite fading channels

Abstract: The symbol-aided (SA) synchronization concept developed by Moher and Lodge (1989) is applied to the MSAT channel modeled with a shadowed Rician process. Simulation data demonstrate that it can track the severe phase jitter encountered on the fading channel free of the false lock which plagues conventional techniques. The algorithm multiplexes known symbols into the data stream, establishing an absolute reference free of decision errors that is used to estimate the fading phase. An improvement to the SA algorithm which extracts phase information from the data-bearing symbols is proposed. It is found that the new technique is more effective for larger K. The improved algorithm is referred to as symbol-aided plus decision-directed (SADD) phase estimation. A system employing SADD phase estimation, trellis-coded modulation, interleaving, and amplitude weighting within the Viterbi decoder yielded the best BER performance on the shadowed MSAT channel considered. >

Direct-sequence spread-spectrum parallel acquisition in nonselective and frequency-selective Rician fading channels

Abstract: This work presents the performance of the direct-sequence spread-spectrum (DS-SS) parallel acquisition system, previously proposed by the authors (1989, 1990), for nonselective and frequency-selective Rician (i.e. specular plus Rayleigh) fading channels. The acquisition system utilizes a bank of parallel I-Q noncoherent matched filters for the search mode, and a coincidence detector for the verification mode. The probabilities of detection and false alarm are derived and the mean and variance of the acquisition time are evaluated as a measure of the system performance. The nonselective channel is a Rayleigh fast fading channel, while the frequency-selective channel model is the so-called wide sense stationary uncorrelated scattering (WSSUS), selective only on frequency. These channels are typical for aircraft-satellite and line-of-sight (LOS) communications. >

A spectrum- and power-efficient EHF mobile satellite system to be integrated with terrestrial cellular systems

Abstract: An EHF satellite system for land-mobile applications to be integrated with a terrestrial cellular system is described. An approach to evaluate the carrier-to-cochannel interference occurring in multispot satellite coverage adopting frequency reuse is introduced, and results from the analysis are shown. Criteria for spectrum efficiency evaluation are also outlined along with traffic and link budget estimates. Possible options for payload implementation and mobile terminal design are presented. >

Spread slotted ALOHA techniques for mobile and personal satellite communication systems

Abstract: The objective is to develop a general theoretical framework for a class of spread slotted ALOHA (SSA) systems. The contributions include: modeling of a generalized spread slotted ALOHA (SSA) system; derivation of computationally efficient closed form expressions for the SSA system throughput and delay taking into account receiver complexity; and presentation of numerical results to validate the derivation as well as to substantiate the superior performance of the proposed scheme. The results demonstrate that SSA schemes offer significant improvements in terms of throughput, delay, and network stability against excessive loading at very acceptable levels of receiver implementation complexities. Also, SSA is found to be highly robust to errors in the time of arrivals and eliminates the need for a guard time. >

A Bayesian maximum-likelihood sequence estimation algorithm for a priori unknown channels and symbol timing

Abstract: It is shown that the optimum demodulator for the case of an a priori unknown channel and symbol timing can be approximated using a modified Viterbi algorithm (VA), in which the branch metrics are obtained from the conditional innovations of a bank of extended Kalman filters (EKFs). Each EKF computes channel and timing estimates conditioned on one of the survivor sequences in the trellis. It is also shown that the minimum-variance channel and timing estimates can be approximated by a sum of conditional EKF estimates, weighted by the VA metrics. Simulated bit error rate (BER) results and averaged-squared channel/timing error trajectories are presented, with estimation errors compared to the Cramer-Rao lower bound. The BER performance of the modified VA is also shown to be superior to that obtained using a decision-directed channel/timing estimation algorithm. >

Perfect N-phase sequences and arrays (spread spectrum communication)

Abstract: Perfect sequences and arrays have periodic autocorrelation functions whose out-of-phase values are zero. Time-discrete N-phase sequences and arrays have complex elements of magnitude one, and one of (2 pi /N)n, 0 >

Two-dimensional modulation codes

Abstract: A new class of run-length-limited codes in introduced. These codes are called two-dimensional or multitrack modulation codes. Two-dimensional modulation codes provide substantial data storage density increases for multitrack recording systems by operating on multiple tracks in parallel. Procedures for computing the capacity of these new codes are given along with fast algorithms for implementing these procedures. Examples of two-dimensional codes are given to provide a comparison between the encoding rates obtainable with multitrack and traditional single-track codes. >

Analysis of a multiple-cell direct-sequence CDMA cellular mobile radio system

Abstract: The performance of a multiple-cell direct-sequence code division multiple-access cellular radio system is evaluated. Approximate expressions are obtained for the area-averaged bit error probability and the area-averaged outage probability for both the uplink and downlink channels. The analysis accounts for the effects of path loss, multipath fading, multiple-access interference, and background noise. Two types of differentially coherent receivers are considered: a multipath rejection receiver and a RAKE receiver with predetection selective combining. Macroscopic base station diversity techniques and uplink and downlink power control are also topics of discussion. >

Techniques for improving the performance of CELP-type speech coders

Abstract: Techniques for improving the performance of CELP (code excited linear prediction)-type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function, which enhances the perceptual quality of the processed speech, is introduced. The combination of harmonic noise weighting and subsample pitch lag resolution significantly improves the coder performance for voiced speech. Strategies for reducing the speech coder's data rate, while maintaining speech quality, are presented. These include a method for efficient encoding of the long-term predictor lags, utilization of multiple gain vector quantizers, and a multimode definition of the speech coder frame. A 5.9-kb/s VSELP speech coder that incorporates these features is described. Complexity reduction techniques which allow the coder to be implemented using a single fixed-point DSP (digital signal processor) are discussed. >

A 128 Kb/s Hi-Fi audio codec based on adaptive transform coding with adaptive block size MDCT

Abstract: A Hi-Fi audio codec with an improved adaptive transform coding (ATC) algorithm is presented using digital signal processors (DSPs). An audio signal with a 20 kHz bandwidth sampled at 48 kHz is coded at a rate of 128 kb/s. The algorithm utilizes adaptive block size selection, which is effective for preecho suppression. A modified discrete cosine transform (MDCT) with a simple window set is employed to reduce block boundary noise without decreasing the performance of transform coding. In addition, a fast MDCT calculation algorithm, based on a fast Fourier transform, is adopted. Weighted bit allocation is employed to quantize the transformed coefficients. The codec was realized by a multiprocessor system composed of newly developed DSP boards. Subjective tests with the codec show that the coding quality is comparable to that of compact disc signals. >

Cyclostationary crosstalk suppression by decision feedback equalization on digital subscriber loops

Abstract: Interference from digital signals in multipair cables has been shown to be cyclostationary under some conditions. This work evaluates the performance of a decision feedback equalizer (DFE) in the presence of cyclostationary interference (CI), intersymbol interference (ISI), and additive white noise (AWN). A comparison between a DFE with CI and one with stationary interference (SI) shows the ability of the DFE to substantially suppress CI. Fractionally spaced and symbol-rate DFE equalizers are also compared and the former is found to yield better performance, especially in the presence of CI. The use of a symbol-rate DFE using an adaptive timing technique that finds the receiver's best sampling phase is proposed for when the fractionally spaced DFE cannot be used because of its complexity. The results also demonstrate the potential benefits of synchronizing central office transmitter clocks, if a fractionally spaced DFE is used at the receiver. >

Error probabilities of fast frequency-hopped FSK with self-normalization combining in a fading channel with partial-band interference

Abstract: Error probability analysis is performed for a binary orthogonal frequency-shift-keying (FSK) receiver using fast frequency-hopped (FFH) spread-spectrum waveforms transmitted over a frequency-nonselective slowly fading channel with partial-band interference. Diversity is performed using multiple hops per data, bit. A nonlinear combination procedure referred to as self-normalization combining is used by the receiver to minimize partial-band interference effects. Diversity is found to completely negate degradation of the self-normalized receiver caused by partial-band interference and offers definite receiver performance improvement when the direct signal component is weak. The self-normalized receiver is sensitive to fading channels. For severe channel fading, the performance of a conventional noncoherent binary FSK receiver is generally either equivalent or superior to that of the self-normalized receiver. >

Hybrid DS/SFH spread-spectrum multiple access with predetection diversity and coding for indoor radio

Abstract: The authors derive close upper and lower bounds on the average bit error probability for hybrid direct-sequence/slow-frequency-hopped spread-spectrum multiple-access (DS/SFH-SSMA) systems with noncoherent DPSK demodulation, using predetection diversity (selection combining and equal gain combining) in conjunction with interleaved channel coding (Hamming