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Problem statement: A fundamental building block for digital communication is the Publickey cryptography systems. Public-Key Cryptography (PKC) systems can be used to provide secure communications over insecure channels without exchanging a secret key. Implementing Public-Key cryptography systems is a challenge for most application platforms when several factors have to be considered in selecting the implementation platform. Approach: The most popular public-key cryptography systems nowadays are RSA and Elliptic Curve Cryptography (ECC). ECC was considered much more suitable than other public-key algorithms. It used lower power consumption, has higher performance and can be implemented on small areas that can be achieved by using ECC. There is no sub exponential-time algorithm in solving the Elliptic curve discrete logarithm problem. Therefore, it offers smaller key size with equivalent security level compared with the other public key cryptosystems. Finite fields (or Galois fields) is considered as an important mathematical theory. Results: Thus, it plays an important role in cryptography. As a result of their carry free arithmetic property, they are suitable to be used in hardware implementation in ECC. In cryptography the most common finite field used is binary field GF ([2.sup.m]). Conclusion: Our design performs all basic binary polynomial operations in Galois Field (GF) using a microcode structure. It uses a bit-serial and pipeline structure for implementing GF operations. Due to its bit-serial architecture, it has a low gate count and a reduced number of I/O pins. Key words: Public-key cryptography, elliptic curve cryptography, Galois field, scalar multiplication, elliptic curve algorithms, INTRODUCTION Public-key cryptography and symmetric-key cryptography are two main categories of cryptography. The Well-known public-key cryptography algorithms are RSA (Rivest et al., 1978), El-Gamal and Elliptic Curve Cryptography. Presently, there [...]

Distributed wireless sensor networks (WSNs) are being proposed for various applications including defense, security, and smart stages. The introduction of hardware wireless sensors in a signal processing education setting can serve as a paradigm for data acquisition, collaborative signal processing, or simply as a platform for obtaining, processing, and analyzing real-life real-time data. In this paper, a software interface that enables the java-digital signal processing (J-DSP) visual programming environment to communicate in a two-way manner with a wireless sensor network is presented. This interface was developed by writing nesC (an extension to the C programming language for sensors) code that enables J-DSP to issue commands to multiple wireless sensor motes, activate specific transducers, and analyze data using any of the existing J-DSP signal processing functions in real time. A series of exercises were developed and disseminated to provide hardware experiences to signals and systems and signal processing undergraduate students. The hardware with the J-DSP software has been used for two semesters in the senior level digital signal processing (DSP) course at Arizona State University. The interface, the exercises, and their assessment (instruments and results) are described in the paper. Index Terms--Digital signal processing (DSP) education, Java, online education, sensors, signal processing, Web-based labs.

Orthogonal frequency-division multiplex (OFDM) offers a low-complexity solution to equalization in multipath channels but does so by increasing the symbol period. This places a limit on the mobility of such systems since time variations in the channel during the symbol period introduce intercarrier interference (ICI), hence, degrading performance. Solutions to reduce ICI in the literature require a high degree of processing. Increasing terminal mobility also places greater requirements on synchronization processing to track the rapidly changing channel. This paper uses multiple antennas at the receiver so that the channel response can be decomposed into a number of more slowly varying channels. Independent synchronization processing and correction can be applied to each of the derived channels before combining the signals prior to the fast Fourier transform (FFT) process. By individually processing the channels, the effective channel is compressed in the time and frequency domains, improving system performance. Perfect tracking of the multipath clusters is initially assumed to show the potential benefits, followed by operation with an idealized tracking algorithm. Operation with more realistic processing algorithms using fixed sectored elements improving the bit error rate (BER) is investigated. Finally, the benefits are then demonstrated with real measured channels from an urban environment. Index Terms--Antenna arrays, digital communications, Digital Video Broadcasting (DVB), orthogonal frequency-division multiplex (OFDM), synchronization.

Constrained sequence codes and pulse shapes are used to control signal characteristics in digital communication systems. This letter outlines a technique to construct codewords and pulse shapes to meet specified spectral constraints. Based on a block-coded representation of the signal, it is shown how symbol values can be evaluated to satisfy the most significant terms of a Fourier series representation of the desired spectral response. Several properties of the resulting signals are given, and examples are presented to confirm the validity of this approach. Index Terms--Constrained sequence codes, pulse shaping, spectral constraints.

In this paper a compact switched-beam antenna is proposed. The antenna is composed of a four-element antenna array based on L-shaped quarter-wavelength slot antenna elements. Such an antenna element is a planar structure and presents a directional radiation pattern in the azimuth plane. Its maximum radiation direction is toward near the direction of the open end of the slot. As a result, the open ends of the four slot antennas are arranged toward 0, [pi]/2, [pi], and 3[pi]/2, respectively. The statuses of these antennas are controlled by some diodes. Consequently, by carefully controlling the diodes, an antenna with several switchable patterns can be achieved. To prove the concept, a 2.4-2.5 GHz switched-beam antenna for WLAN applications is designed and implemented. Its size is 52 mm in square. The antenna possesses eight directional patterns and many nearly omnidirectional patterns in the azimuth plane. The experiment results fully demonstrate the performance of the proposed design. The envelope correlations and the characteristics of the designed antenna are also discussed. Due to the compact size and low manufacture cost, such a design can be a promising solution for digital home applications to overcome multipath problems and increase the transmission data rate. Index Terms--Antenna array, correlation, slot antenna.

In this paper we first perform the density evolution of LDPC codes in layered multi-antenna architectures using bit-interleaved coded modulation and detection based on successive decoding and interference cancellation. The noise thresholds of LDPC codes are shown to be achieved within 1.2 dB of the capacity for these architectures. We then present a rate allocation method which optimizes the performance of a layered architecture and find good degree distributions for LDPC codes with the rates allocated by our method. Simulation results show that the performance for an optimized layered architecture is better by 4 dB than that for the layered architecture using LDPC codes with the same rate in all layers. Index Terms--Density evolution, layered architecture, low-density parity-check code, multiple antennas.

A system of functions (signals) on the finite line, called the oscillator system, is described and studied. Applications of this system for discrete radar and digital communication theory are explained. Weil representation | commutative subgroups | eigenfunctions | random behavior | deterministic construction

A single-carrier two-dimensional modulation scheme operating over a linear time-dispersive channel is described. The synthesis of the widely linear (WL) receiver is performed by assuming perfect knowledge of the in-phase and quadrature-phase (IQ) imbalance parameters, but a new blind algorithm for IQ imbalance parameters is also proposed.

A factorization based approach is proposed for estimating channel impulse responses. The factorization based approach can work with a small number of samples and has become more efficient than the subspace method when the length of packets is long.

A novel self-timed communication protocol is based on the phase-modulation of a reference signal. The reference signal is sent on a number of transmission lines and the data can be recovered observing the sequence of events on the these lines. Employing several fines increases the number of states hence reducing the number of symbols required for a transmission. A new encoding algorithm is described which generates symbol-dependent matrices which are used to control the phase of transmission lines. The protocol concept, the algorithm and analysis of the system, together with simulation results, are presented. Index Terms--Digital communication, encoding, fault tolerance, interconnects circuit interconnect.

In digital communication systems, variable-length Walsh codes, also called orthogonal variable spreading factor codes, are used to support simultaneous variable-rate data transmission. This paper describes algorithms for dynamic variable-length Walsh code assignment. Optimal assignment criterion is derived and an efficient data structure for maintaining variable-length Walsh codes is introduced. The benefits of code reallocation are also studied. Performance is evaluated via simulation. Index Terms--Walsh code, orthogonal variable spreading factor (OVSF) code, orthogonal channelization code, maximum-weight criterion, variable data rate communications.

Data converters, in particular, the transmit digital-to-analog converter (DAC), should not limit performance in a full-duplex digital-communication transceiver. Typically, the number of DAC bits is chosen to be large enough so that the effect of DAC quantization noise on the local receiver is negligibly small. As described in this brief, the DAC quantization noise can be cancelled in an echo-canceling full-duplex transceiver. The proposed quantization-noise cancellation allows use of a simpler, lower resolution DAC. The quantization-noise cancellation concept is described, and simulation results are presented that demonstrate its operation. Index Terms--Analog front-end (AFE), digital communication, digital-to-analog converter (DAC), echo cancellation, noise cancellation.

Our purpose is to find an upper hound for the length of an s-extremal code over [F.sub.2] (resp. [F.sub.4]) when d [equivalent to] 2 (mod 4) (resp. d odd). This question is left open in [A bound for certain s-extremal lattices and codes, Archiv der Mathematik, vol. 89, no. 2, pp. 143-151, 2007] (resp. [s-extremal additive [F.sub.4] codes, Advances in Mathematics of Communications, vol. 1, no. 1, pp. 111-130, 2007]). More precisely, we show that if there is an [n, n/2, d] s-extremal Type I binary self-dual code with d > 6 and d [equivalent to] 2 (mod 4), then n < 21d - 82. Similarly we show that if there is an (n, [2.sup.n], d) s-extremal Type I additive self-dual code over [F.sub.4] with d > 1 and d [equivalent to] 1 (mod 2), then n < 13d - 26. We also define s-extremal self-dual codes over [F.sub.2] + u[F.sub.2] and derive an upper bound for the length of an s-extremal self-dual code over [F.sub.2] + u[F.sub.2], using the information on binary s-extremal codes. Index Terms--Additive [F.sub.4] code, code over [F.sub.2] + u[F.sub.2], s-extremal code, self-dual code.

A generic and novel distribution, referred to as N*Nakagami, constructed as the product of N statistically independent, but not necessarily identically distributed, Nakagami-m random variables (RVs), is introduced and analyzed. The proposed distribution turns out to be a very convenient tool for modelling cascaded Nakagami-m fading channels and analyzing the performance of digital communications systems operating over such channels. The moments-generating, probability density, cumulative distribution, and moments functions of the N*Nakagami distribution are developed in closed form using the Meijer's G-function. Using these formulas, generic closed-form expressions for the outage probability, amount of fading, and average error probabilities for several binary and multilevel modulation signals of digital communication systems operating over the N*Nakagami fading and the additive white Gaussian noise channel are presented. Complementary numerical and computer simulation performance evaluation results verify the correctness of the proposed formulation. The suitability of the N*Nakagami fading distribution to approximate the lognormal distribution is also being investigated. Using Kolmogorov-Smirnov tests, the rate of convergence of the central limit theorem as pertaining to the multiplication of Nakagami-m RVs is quantified. Index Terms--Applied stochastic models, bit error rate (BER), cascaded fading, central limit theorem (CLT), keyhole channels, Kolmogorov-Smirnov test, lognormal fading, multiple-input multiple-output (MIMO), Nakagami-m, outage probability (OP).

A high-directivity combined self-beam/null-steering array for secure point-to-point binary phase-shift keying communications is introduced. The system provides high directivity and reduced probability of interception using just two antenna elements. Using quadrature phase-shift keying modulators allows for compact single-layer fabrication. The 2.4-GHz prototype is tested at interrogation angles of 0[degrees], -10[degrees], and +20[degrees], and demonstrates high signal-to-interference ratio directivity, completely disabling interception [+ or -]20[degrees] from the direction of the interrogator. The system should find various applications where secure communications are required. Index Terms--Digital communication, microwave receivers, phase conjugation, phased arrays, transponders. Digital Object Identifier 10.1109/TMTT.2007.895405

We present information-theory analysis of the tradeoff between bit-error rate improvement and the data-rate loss using skewed channel coding to suppress pattern-dependent errors in digital communications. Without loss of generality, we apply developed general theory to the particular example of a high-speed fiber communication system with a strong patterning effect. Index Terms--Digital communication, error correction, intersymbol interference (ISI), information theory.

This paper presents the performance analysis of a conditional diversity combining sebeme, which is called output-threshold maximum ratio combining (OT-MRC), over the independent identically distributed Nakagami-m with integer m and independent hut not necessarily identically distributed Rayleigh fading channels. Based on the closed-form expressions for the statistics, the authors analyze the performance of the OT-MRC systems and compare it with that of the conventional MRC systems. They exploit the tradeoff between complexity and performance in this scheme over the Nakagami fading channels and see the effect of unbalance of the fading gains over multiple diversity branches on the performance of OT-MRC. They show, from some numerical examples, that the OT-MRC scheme, when compared with the MRC scheme, can reduce the complexity significantly without losing the required performance. Index Terms--Diversity combining, Nakagami-m fading channels, nonindependent identically distributed (i.i.d.) Rayleigh fading channels, output-threshold maximum ratio combining (OT-MRC).

Several digital cellular systems employ coherent Rake reception, wherein channel estimates are used to combine despread values. In this paper, a maximum likelihood (ML) approach to Rake combining that accounts for channel estimation error, including error correlation across despread values, is examined. Fading and noise correlation are also considered. The performance of the ML approach is compared to traditional Rake combining as well as approximate ML-based approaches. Results show that when the channel estimation error and noise are of the same order, ML-based approaches can provide gains on the order of 1 dB over traditional Rake reception. Index Terms--Channel estimation, code division multiple access (CDMA), coherent demodulation, demodulation, maximum likelihood detection, multipath channels, spread spectrum communication.

In the design of RF up-conversion and down-conversion communication links, an issue of special interest is presented by the nonlinear characteristic of analog devices. In this paper, we deal with the effect of memoryless nonlinear distortion on orthogonal frequency-division multiplexing (OFDM) transceivers. We tackle the issue of calculation of the number of intermodulation products with methods from combinatorics theory and derive closed-form expressions for the signal-to-noise ratio (SNR). We deal with third-order nonlinearities alone though the methodology used can be extended to cover higher order nonlinear phenomena. We then proceed to deriving SNR expressions in the presence of a high adjacent channel of the same service and predict the generation of in-band tonal interference. Finally, we generalize to the case of a multichannel OFDM transceiver. In each case, bit-error-rate estimations for differential quadrature phase-shift keying and symbol-error-rate estimations for hi-quadrature amplitude-modulation constellations are presented and a mapping between circuit characteristics and OFDM performance is outlined. Index Terms--Adjacent channel, bit error rate (BER), combinatorics, digital audio broadcasting (DAB), digital video broadcasting (DVB), discrete multitone (DMT), intermodulation, multicarrier system, nonlinearity, orthogonal frequency-division multiplexing (OFDM), RF circuits, signal-to-noise ratio (SNR), symbol error rate (SER).

This paper introduces two different iterative receivers for space--time-coded data transmitted through frequency-selective fading channels. Both receivers perform maximum a posteriori data detection and decoding, and operate semiblindly by making use of the expectation maximization (EM) algorithm to estimate the unknown channel. The first receiver utilizes a supertrellis to jointly describe the channel and the code, while in the second one, this description is accomplished with two separated trellises. The two approaches are compared in terms of trellis complexity, achievable diversity level, and bit error rate (BER). The supertrellis approach is less computationally demanding and has a faster convergence rate, while the separate approach can achieve a higher diversity level and thus a smaller BER. This tradeoff is explored with examples and computer simulations, which show that, in many cases, the supertreilis approach can be a better solution than the separate approach. Index Terms--Combined estimation, detection and decoding, expectation maximization (EM) algorithm, space-time coding.

In this paper, a new modulation scheme that is called spectrally shaped generalized multitone direct sequence spread spectrum (SSG-MT-DS-SS) is proposed. This scheme enables moderate shaping of the transmitted spectrum without the use of spectral-shaping filters; instead, the spectrum is shaped by appropriate selection of the parameters of each subcarrier. These parameters are amplitude, data rate, processing gain, and the frequency separation between adjacent subcarriers. Since these parameters are selectable, this scheme is a completely generalized MT-DS-SS approach. Analysis and simulation results show that by properly choosing these parameters, the level of the spectral sidelobes can be reduced, while slightly changing the width of the mainlobe and simultaneously obtaining a good error-probability performance, on both additive white Gaussian noise and dispersive fading channels. For even a small number of subcarriers (e.g., 3-5), spectral sidelobes can be reduced by 3-6 dB from those of the conventional MT-DS-SS spectrum, with little or no degradation in error-probability performance. For larger numbers of subcarriers, additional sidelobe reductions are attainable. In addition, the SSG-MT-DS-SS system has a smaller peak-to-average power ratio than the conventional MT-DS-SS when the amplitudes of each subcarrier are not identical. Index Terms--Bit error ratio (BER) performance, multitone, spectrum shaping, spread spectrum (SS).

A method is presented for retransmitting a digital communication signal over channels with different signal-to-noise ratios (SNRs), without complete demodulation and remodulation (Demod/Remod). The method, which we call signal code modulation (SCM), converts samples of the incoming signal into digital and analog symbols whose characteristics are designed to match the channel. The SCM method trades bandwidth for SNR in the classical manner and achieves a performance close to that of conventional Demod/Remod methods. However, it does so by performing relatively simple operations on the sampled input signal and does not require detailed knowledge of its fine structure (framing, coding, etc.). Index Terms--Bit error rate (BER), coding, digital communications, quadrature amplitude modulation (QAM), signal code modulation (SCM), symbol error rate (SER).

A study presents a new paradigm for the construction of reversible transforms that map integers to integers. Modulo transforms present a means of achieving reversible integer transforms and an alternative to lifting in certain situations.

Starting with the maximum a posteriori (MAP) estimation approach, this paper derives the optimum (in the MAP estimation sense) means for performing symbol-timing recovery in the absence of carrier-phase information (i.e., prior to carrier-phase recovery). Specifically, we examine the necessary modification of a well-known form of coherent symbol synchronizer, namely, the data transition tracking loop (DTTL), to allow its operation in the absence of carrier-phase information, i.e., as a so-called noncoherent symbol sync loop. By employing such a noncoherent scheme, one can eliminate the need for iteration between the carrier and symbol sync functions, as typically takes place in receivers that more commonly perform carrier tracking and acquisition prior to symbol timing. The performance of both the linear and nonlinear versions of this noncoherent DTTL is obtained by a combination of analysis and simulation, and compared with that of the corresponding coherent DTTLs. Index Terms--Data transition tracking loop (DTTL), noncoherent symbol sync loop, symbol synchronization.

An iterative joint finite-impulse response (FIR) zero-forcing (ZF) precoder-equalizer optimizaion algorithm for multiple-input multiple-output intersymbol interference (MIMO-ISI) channel is proposed. The existing joint precoder-equalizer design algorithms for MIMO-ISI channels require a guard period, which is unnecessary redundancy consuming the valuable channel bandwidth.

Multirate (MR) modulation resembles block-coded modulation (BCM), since matrices are being used to transform binary input vectors to multilevel output vectors (blocks) of length K. Unlike BCM, attention is given to the spectral shaping of the signal to be transmitted. Hence, the encoding matrices are designed to provide simultaneous spectral shaping and Euclidean distance. The encoding matrices can be implemented by using MR digital filters of low complexity. MR modulation also resembles partial response (PR) modulation since, in both cases, a transmitter and receiver filter is used with an overall duobinary impulse response. It will be shown that MR modulation has a number of significant advantages compared with PR modulation. Thus, for example, with MR modulation, loss of synchronization or gain control, as can occur with PR modulation, cannot happen in the receiver. Furthermore, computer simulations for an additive white Gaussian noise channel demonstrate that, for a bit-error rate of [10.sup.-6], MR modulation (with K = 10) gives a gain of 1.5 dB, compared with PR modulation and symbol-by-symbol detection. However, MR modulation requires a slightly higher bandwidth. It is also explained how, for block lengths K [greater than or equal to] 10, MR modulation gives a larger bandwidth efficiency than M-ary pulse-amplitude modulation with raised-cosine pulses and a rolloff factor [alpha] [greater than or equal to] 0.1. Index Terms--Bandlimited communication, decoding, digital communication, discrete-time filters, modulation.

Abstract: Digital communications systems use Mufti tone Channel (MC) transmission techniques with differentially encoded and differentially coherent demodulation. Today there are two principle MC application, one is for the high [...]

This paper presents the physical layer functions powering a new wireless data broadcasting system over analog television signals, referred to as dNTSC system (for data over National Television System Committee (NTSC) standard). Novel data insertion techniques at the transmitter and data extraction techniques at the receiver allow data rates of 1-4 Mbps that coexist with existing analog transmission but do not adversely affect normal television reception. Novel digital signal processing (DSP) techniques include a video abatement system that reduces data-to-video crosstalk for unimpaired analog television reception, adaptive nonlinear amplifier compensation, and advanced video cancellation and adaptive data equalization methods in the receiver. An analysis of the dNTSC system through noisy, multipath channels reveals the subtleties of the dNTSC system, compared to a conventional, dedicated digital communication link. Index Terms--Communication channel, digital communication, digital data broadcast, equalization, receiver, transmitter.

Signal fading is a ubiquitous problem in mobile and wireless communications. In digital systems, fading results in bit errors, and evaluating the average error rate under fairly general fading models and multichannel reception is often required. Predominantly to date, most researchers perform the averaging using the probability density function method or the moment generating function (MGF) method. This paper presents a third method, called the characteristic function (CHF) method, for calculating the average error rates and outage performance of a broad class of coherent, differentially coherent, and noncoherent communication systems, with or without diversity reception, in a myriad of fading environments. Unlike the MGF technique, the proposed CHF method (based on Parseval's theorem) enables us to unify the average error-rate analysis of different modulation formats and all commonly used predetection diversity techniques (i.e., maximal-ratio combining, equal-gain combining, selection diversity, and switched diversity) within a single common framework. The CHF method also lends itself to the averaging of the conditional error probability involving the complementary incomplete Gamma function and the confluent hypergeometric function over fading amplitudes, which heretofore resisted to a simple form. As an aside, we show some previous results as special cases of our unified framework. Index Terms--Characteristic function (CHF) method, digital communications, fading channels, frequency-domain analysis, wireless communications.

The expected complexity of sphere decoding that is applied to a large class of problems is exponential in the number of symbols jointly detected after obtaining a lower bound. An increase in signal-to-noise ratio (SNR) reduces the complexity of sphere decoding.

The bidirectional arbitrated decision-feedback equalizer (BAD), which has bit-error rate performance between a decision-feedback equalizer (DFE) and maximum a posteriori (MAP) detection, is presented. The computational complexity of the BAD algorithm is linear in the channel length, which is the same as that of the DFE, and significantly lower than the exponential complexity of the MAP detector. While the relative performance of BAD to those of the DFE and the MAP detector depends on the specific channel model, for an error probability of [10.sup.-2], the performance of BAD is typically 1-2 dB better than that of the DFE, and within 1 dB of the performance of MAP detection. Index Terms--Arbiter, decision-feedback equalizers (DFEs), digital communications, equalizers, multipath channels, time-reversal diversity.

The problem of a posteriori joint detecting and discriminating pulses in a quasiperiodic pulse train is studied. An efficient computing algorithm for the exact solution to this problem is derived and proved for an exact solution to the problem.

A wireless sensor server is developed based on a reconfigurable active smart-antenna/retrodirective array. The system can serve as both a retrodirective array transponder and a smart-antenna receiver simply by changing the frequency of the local oscillator applied to the mixers, enabling it to best utilize its hardware to suit its communication environment. When operating as a direct-conversion receiver, the receiver array successfully demodulates a quaternary phase-shift keying (QPSK) modulated signal with circuit gain of 7 dB and [E.sub.b]/[N.sub.O] for BER = [10.sup.-4] is approximately 12 dB without any error correction. In the retrodirective array mode, the system provides 20-dB circuit gain and 20-dB radio-frequency-intermediate-frequency isolation at the center frequency as well as phase conjugation, exhibiting excellent retrodirectivity. The mixers perform phase conjugation and modulation simultaneously, enabling the transmission of locally stored data. The local data is successfully extracted by an interrogator. Index Terms--Digital communication, microwave receivers, phase conjugation, phased arrays, transponders.

We designed a multi-channel Nakagami fading simulator by modeling the received combined signal-to-noise ratio as a finite-state Markov chain, following a previously proposed approach. Our model generates directly the error process at the output of a diversity receiver and can emulate selection, maximal-ratio, and equal-gain combining. As the order of diversity increases, the savings in computational complexity improve linearly with respect to a traditional waveform simulator. The level crossing rates of the simulated envelope are shown to be very close to their theoretical values. The simulator's performance is also evaluated in terms of the accuracy of the obtained bit error rates, for both uncoded and coded systems. The simulator speeds up the performance evaluation of high-rate communication links where a high number of samples is needed. Index Terms--Diversity methods, fading channels, level-crossing problems, radio broadcast receivers, simulation software.

The combination of high capacity, signal robustness, and worldwide acceptance have made digital terrestrial broadcasting (DVB-T) one of today's more promising technologies, while newly adopted standards support the integration of digital television programs and data services within a single multiplex, offered to both stationary and mobile end users. This paper addresses the issue of optimal bit rate allocation among DTV and IP services within the DVB-T multiplex, and demonstrates a novel configuration providing real-time sharing of the available bandwidth via an algorithm based on service hierarchy and perceptual quality assessment. The architecture and performance of a fully functional DVB-T testbed developed on this concept is also presented. Index Terms--Dynamic bandwidth allocation, IP over DVB, picture quality assessment.

The high peak-to-average power ratio (PAR) in Orthogonal Frequency Division Multiplexing (OFDM) modulation systems can significantly reduce power efficiency and performance. Methods exist which alter or introduce new signal constellations to combat large signal peaks. We present a new PAR-reduction method that dynamically extends outer constellation points in active (data-carrying) channels, within margin-preserving constraints, in order to minimize the peak magnitude. This scheme simultaneously decreases the bit error rate slightly while substantially reducing the peak magnitude of an OFDM transmit block. Furthermore, there is no loss in data rate and unlike other methods, no side information is required. PAR reduction for an approximated analog signal is considered, and about a 4.6 dB reduction at a [10.sup.-5] symbol-clip probability is obtained for 256-channel QPSK OFDM. The results show great promise for use in commercial systems. Index Terms--Multicarrier communications, OFDM, PAR reduction, peak power reduction, peak-to-average ratio.

The problem of allocating network resources to application sessions backlogged at an individual switch has a great impact on the end-to-end delay and throughput guarantees offered by the network. There exists a class of algorithms based on Weighted Fair Queueing (WFQ) for scheduling packets which are work-conserving and they guarantee fairness to the backlogged sessions. These algorithms also apply to ATM networks with packet equal to a single cell or an ATM block of (fixed size). Bursts are groups of varying numbers of cells. We generalize WFQ to schedule bursts. Our motivation is to derive an adaptive algorithm which generalizes the (fixed size) packet level to a varying size packet level. The new algorithm enhances the performance of the switch service for many important applications. The proposed scheme maintains the work-conserving property, and also provides throughput and fairness guarantees. The worse-case delay bound is also given. We use simulation to study the performance characteristics of our algorithm. Our results demonstrate the efficiency of the new algorithm. Index Terms--ATM networks, cell burst, fair queueing, quality of service (QoS), quality measurement unit.

High-speed broadband digital communication networks rely on digital multiplexing technology where clock synchronization, including processing, transmission, and recovery of the clock, is the critical technique. This paper interprets the process of clock synchronization in multiplexing systems as quantizing and Coding the information of clock synchronization, interprets clock justification as timing sigma-delta modulation (T[DELTA]--[SIGMA]M), and interprets the jitter of justification as quantization error. As a result, decreasing the quantization error is equivalent to decreasing the jitter of justification. Using this theory, the paper studies the existing jitter-reducing techniques in transmitters and receivers, presents some techniques that can decrease the quantization error (justification jitter) in digital multiplexing systems, and presents a new method of clock recovery. Index Terms--Clock recovery, clock synchronization, digital multiplexing, jitter, T[DELTA]--[SIGMA]M.

In synchronous digital hierarchy and plesiochronous digital hierarchy networks, it is frequently necessary to recover a data clock from a gapped clock derived from stuff information present at the desynchronizer. In this paper, a comprehensive analysis of the timing jitter resulting from phase-locked loop-type desynchronizers is presented. This analysis is different from the conventional analysis where the jitter is represented using a phase-error sequence. It is shown that such a simplified approach cannot accurately describe the jitter at the output of the desynchronizer. From the detailed analysis, it is also shown how the use of threshold modulation at the synchronizer reduces the low-frequency jitter at the desynchronizer. It has been demonstrated in the paper that when threshold modulation is used at the synchronizer, the dominating low-frequency jitter terms cannot be explained by the conventional jitter analysis methods. Therefore, in future networks, where tighter jitter performances are to be imposed on the synchronizers, jitter characterization using the proposed true jitter analysis technique would be very useful. Index Terms--Digital communications, jitter, SONET, synchronization, synchronous digital hierarchy (SDH).

Using a circular contour integral representation for the generalized Marcum-Q function, [Q.sub.m] (a, b), we derive a new closed-form formula for the moment generating function (MGF) of the output signal power of a dual-diversity selection combiner (SC) in bivariate (correlated) Nakagami-m fading with positive integer fading severity index. This result involves only elementary functions and holds for any value of the ratio a/b in [Q.sub.m] (a, b). As an aside, we show that previous integral representations for [Q.sub.m] (a, b) can be obtained from a contour integral and also derive a new, single finite-range integral representation for [Q.sub.m] (a, b). A new infinite series expression for the MGF with arbitrary m is also derived. These MGFs can be readily used to unify the evaluation of average error performance of the dual-branch SC for coherent, differentially coherent, and noncoherent communications systems. Index Terms--Digital communications, diversity reception, fading channels, wireless communications.

The Nuttall Q function is introduced and, for the case where the sum of its two order indices is odd, a closed-form expression is derived for it in terms of the Marcum Q function and modified Bessel functions of the first kind. Alternatively, a finite-range integral expression is also given for it, which has the identical form to that of the generalized Marcum Q function. Finally, digital communication performance applications involving the Nuttall Q function are presented to justify the value of the results. Index Terms--Marcum Q function, Nuttall Q function.

In this brief, simple noncoherent detection methods for chaos-shift-keying (CSK) modulation are proposed. By exploiting some deterministic property of the two chaotic maps, the proposed methods recover the digital message through simple decision algorithms. Specifically, the proposed methods exploit the difference in the return maps of the signals representing the digital symbols. Two specific algorithms are discussed, namely a regression-based algorithm and a probability-based algorithm. Simple tent maps are used for illustration. The bit-error-rate under additive white Gaussian noise is studied by computer simulations. Index Terms--Chaos-based communication, digital communication, error probability, least-squares regression, noncoherent detection, return maps.

The Kahn envelope elimination and restoration (EER) technique allows for linear RF power amplification by combining nonlinear, but efficient, RF and AF power amplifiers (PAs). In order to use the EER technique for digital signals, a coordinate transform from the original Cartesian in-phase and quadrature mode into a polar mode has to take place, yielding an envelope (or amplitude) and a PM RF signal. This coordinate transform is extremely nonlinear and thereby broadens the spectra of the original signals. In the final PA stage, both signals are recombined. However, since this recombination process is imperfect, out-of-band (OOB) emissions come up, also known as adjacent channel power or spectral regrowth. In this paper, the impact of the broadening of the amplitude and phase signals on OOB emissions is investigated with respect to imperfect restoration due to signal delays and limited bandwidth of the amplitude path. It is shown that the amount of OOB emissions can significantly be reduced if the modulation scheme shows a 'hole' at the origin in its vector diagram. Index Terms--ACPR, amplifier, band-limited communications, delay effects, EER, Kahn technique, PM, spectral regrowth, transmitter.

We have suggested bit-interleaved coded modulation with soft decision iterative decoding (BICM-ID) for bandwidth-efficient transmission over Gaussian and fading channels. Unlike trellis coded modulation, BICM-ID has a small free Euclidean distance but large diversity order due to bit interleaving. With iterative decoding, soft bit decisions can be employed to significantly improve the conditional intersignal Euclidean distance. This leads to a large coding gain, comparable to that of turbo TCM, over both Gaussian and Rayleigh fading channels with much less system complexity. We address critical design issues to enhance the decoding performance and provide the analytical bounds on the performance with an ideal feedback assumption. We investigate the performance characteristics of BICM-ID through extensive simulations and show that at high signal to noise ratios, the performance of BICM-ID converges to the performance assuming error-free feedback. Index Terms--BICM, coded modulation, digital communications, iterative decoding, turbo codes.