M.L.R. de Campos

Partial-update NLMS algorithms with data-selective updating

In this paper, we present mean-squared convergence analysis for the partial-update normalized least-mean square (PU-NLMS) algorithm with closed-form expressions for the case of white input signals. The formulae presented here are more accurate than the ones found in the literature for the PU-NLMS algorithm. Thereafter, the ideas of the partial-update NLMS-type algorithms found in the literature are incorporated in the framework of set-membership filtering, from which data-selective NLMS-type algorithms with partial-update are derived. The new algorithms, referred to herein as the set-membership partial-update normalized least-mean square (SM-PU-NLMS) algorithms, combine the data-selective updating from set-membership filtering with the reduced computational complexity from partial updating. A thorough discussion of the SM-PU-NLMS algorithms follows, whereby we propose different update strategies and provide stability analysis and closed-form formulae for excess mean-squared error (MSE). Simulation results verify the analysis for the PU-NLMS algorithm and the good performance of the SM-PU-NLMS algorithms in terms of convergence speed, final misadjustment, and computational complexity.

Spectrally Efficient UWB Pulse Shaping With Application in Orthogonal PSM

In this paper, we present a method to obtain a set of orthogonal pulses to be used in pulse-shape modulation (PSM) for ultra-wideband communications. The pulses are built as linear combinations of Hermite functions, which are shown to have unique advantageous features. Mathematical restrictions of orthogonality and spectral efficiency are introduced as guidelines to a fully explained search procedure to find the best set of pulses. Additionally, this procedure is adapted and used to find a single FCC-compliant pulse shape. A quaternary PSM scheme is implemented with orthogonal pulses obtained by the proposed method, and the results of a simulation are shown

Constrained adaptation algorithms employing Householder transformation

This paper presents a tutorial-like detailed explanation of linearly constrained minimum-variance filtering in order to introduce an efficient implementation that utilizes Householder transformation (HT). Through a graphical description of the algorithms, further insight on linearly constrained adaptive filters was made possible, and the main differences among several algorithms were highlighted. The method proposed herein, based on the HT, allows direct application of any unconstrained adaptation algorithm as in a generalized sidelobe canceller (GSC), but unlike the GSC, the HT-based approach always renders efficient implementations. A complete and detailed comparison with the GSC model and a thorough discussion of the advantages of the HT-based approach are also given. Simulations were run in a beamforming application where a linear array of 12 sensors was used. It was verified that not only the HT approach yields efficient implementation of constrained adaptive filters, but in addition, the beampatterns achieved with this method were much closer to the optimal solution than the beampatterns obtained with GSC models with similar computational complexity.

Low-complexity constrained affine-projection algorithms

This paper proposes low-complexity constrained affine-projection (CAP) algorithms. The algorithms are suitable for linearly constrained filtering problems often encountered in communications systems. The CAP algorithms derived in this paper trade convergence speed and computational complexity in the same way as the conventional affine-projection (AP) algorithm. In addition, data-selective versions of the CAP algorithm are derived based on the concept of set-membership filtering. The set-membership constrained affine-projection (SM-CAP) algorithms include several constraint sets in order to construct a space of feasible solutions for the coefficient updates. The SM-CAP algorithms include a data-dependent step size that provides fast convergence and low mean-squared error. The paper also discusses important aspects of convergence and stability of constrained normalized adaptation algorithms and shows that normalization may introduce bias in the final solution.

The constrained conjugate gradient algorithm

Based on the condition for equivalence between linearly constrained minimum-variance (LCMV) filters and their generalized sidelobe canceler (GSC) implementations, we derive the new constrained conjugate gradient (CCG) algorithm. We discuss the use of orthogonal and nonorthogonal blocking matrices for the GSC structure and how the choice of this matrix may affect the relationship with the LCMV counterpart. The newly derived algorithm was tested in a computer experiment for adaptive multiuser detection and showed excellent results.

On the equivalence of RLS implementations of LCMV and GSC processors

This letter compares the transients of the constrained recursive least squares (CRLS) algorithm with the generalized sidelobe canceler (GSC) employing the recursive least squares (RLS) algorithm. We prove that the two adaptive implementations are equivalent everywhere regardless of the blocking matrix chosen. This guarantees that algorithm tuning is not affected by the blocking matrix. This result differs from the more restrictive case for transient equivalence of the constrained least mean-square (CLMS) algorithm and the GSC employing the least mean square (LMS) algorithm, for in this case the blocking matrix needs to be unitary.

Jointly minimum BER transmitter and receiver FIR MIMO filters for binary signal vectors

A theory is developed for jointly minimizing the bit error rate (BER) between the desired and decoded signals with respect to the coefficients of transmitter and receiver finite impulse response (FIR) multiple-input multiple-output (MIMO) filters. The original signal is assumed to be a vector time-series with equally likely memoryless Bernoulli vector components. The channel model constitutes of a known FIR MIMO transfer function and Gaussian additive noise independent of the original signal. The channel input signal is assumed to be power constrained. Based on the formulas obtained, an iterative numerical optimization algorithm is proposed. When compared with other design methods available in the literature, the proposed method yields better results due to the generality of the model considered and the joint optimization of the transmitter-receiver pair.

Performance comparison of binary and quaternary UWB modulation schemes

In this paper we test and compare different modulation strategies to be used in ultra-wide band (IR-UWB) communications. In the UWB systems of interest in this work the information is conveyed by short-duration pulses; the modulation scheme determine how the data stream is to be transmitted over those pulses. Here we test and compare several schemes based on three types of modulation. The first type is the pulse position modulation (PPM), which includes a pulse delay according to the data to be transmitted. The second is the recently proposed pulse shape modulation (PSM), that uses a different pulse shape to each data. The third one, proposed in this paper, is an M-ary modulation scheme that combines the shape and amplitude of the pulse to transmit the data, which we call pulse amplitude and shape modulation (PASM). Other M-ary schemes are tested, including the recent orthogonal M-ary PSM, based on orthogonal Hermite functions, and a new quaternary PPM scheme, also proposed here. All schemes are tested over an AWG channel.In this paper we test and compare different modulation strategies to be used in ultra-wide band (IR-UWB) communications. In the UWB systems of interest in this work the information is conveyed by short-duration pulses; the modulation scheme determine how the data stream is to be transmitted over those pulses. Here we test and compare several schemes based on three types of modulation. The first type is the pulse position modulation (PPM), which includes a pulse delay according to the data to be transmitted. The second is the recently proposed pulse shape modulation (PSM), that uses a different pulse shape to each data. The third one, proposed in this paper, is an M-ary modulation scheme that combines the shape and amplitude of the pulse to transmit the data, which we call pulse amplitude and shape modulation (PASM). Other M-ary schemes are tested, including the recent orthogonal M-ary PSM, based on orthogonal Hermite functions, and a new quaternary PPM scheme, also proposed here. All schemes are tested over an AWG channel.

Speech privacy for modern mobile communication systems

Speech privacy techniques are used to scramble clear speech into an unintelligible signal in order to avoid eavesdropping. Some analog speech-privacy equipments (scramblers) have been replaced by digital encryption devices (comsec), which have higher degree of security but require complex implementations and large bandwidth for transmission. However, if speech privacy is wanted in a mobile phone using a modern commercial codec, such as the AMR (adaptive multirate) codec, digital encryption may not be an option due to the fact that it requires internal hardware and software modifications. If encryption is applied before the codec, poor voice quality may result, for the vocoder would handle digitally encrypted signal resembling noise. On the other hand, analog scramblers may be placed before the voice encoder without causing much penalty to its performance. Analog scramblers are intended in applications where the degree of security is not too critical and hardware modifications are prohibitive due to its high cost. In this article we investigate the use of different techniques of voice scramblers applied to mobile communications vocoders. We present our results in terms of LPC and cepstral distances, and PESQ values.

Time-Varying FIR Transmultiplexers With Minimum Redundancy

In this paper, we derive conditions for existence of zero-forcing equalizers for FIR transmultiplexers. We extend theoretical results from the literature for zero-forcing equalization in transmultiplexer systems, and derive new conditions for a more general configuration that includes filter-bank based systems with long responses and time-varying filter banks. The time-varying filter banks can be used to model code division multiple access systems with long codes, as well as other practical problems. The results can be applied to both downlink and uplink scenarios, due to the general framework considered for the analysis. The derived relations allow the use of relatively simple equalizers, lead to transmitters using small amount of redundancy, and also allow the transmission through channels with long impulse responses. Experimental results obtained via computer simulations validate the derived expressions. The simulations were carried out for zero-forcing design and also for least-squares and minimum mean squared error designs that use the relations derived for existence of zero-forcing equalizers. The results show that the performance of all equalizers improve if the zero-forcing conditions derived in this paper are followed.