Sergio L. Netto

Adaptive IIR filtering algorithms for system identification: a general framework

Adaptive IIR (infinite impulse response) filters are particularly beneficial in modeling real systems because they require lower computational complexity and can model sharp resonances more efficiently as compared to the FIR (finite impulse response) counterparts. Unfortunately, a number of drawbacks are associated with adaptive IIR filtering algorithms that have prevented their widespread use, such as: convergence to biased or local minimum solutions; requirement of stability monitoring; and slow convergence. Most of the recent research effort on this field is aimed at overcoming some of the above mentioned drawbacks. In this paper, a number of known adaptive IIR filtering algorithms are presented using a unifying framework that is useful to interrelate the algorithms and to derive their properties. Special attention is given to issues such as the motivation to derive each algorithm and the properties of the solution after convergence. Several computer simulations are included in order to verify the predicted performance of the algorithms. >

On the design of high-complexity cosine-modulated transmultiplexers based on the frequency-response masking approach

Two efficient techniques exploiting the frequency-response masking (FRM) approach are proposed in order to make it feasible to design prototype filters for highly selective nearly perfect-reconstruction cosine-modulated transmultiplexers and filter banks (CMTs and CMFBs) having a very large number of channels. In these design schemes, the number of unknowns is drastically reduced when compared with the corresponding techniques for designing direct-form prototype filters. Furthermore, in the proposed techniques, the main figures of merits, that is, the intersymbol interference and the interchannel interference for CMTs and the overall and aliasing distortions for CMFBs are taken into account in a controlled manner. In order to speed up the convergence of these two optimization techniques, simplifications for computing the resulting nonlinear constraints and the corresponding gradient vectors are proposed. They differ from each other in the sense that the first and second ones utilize the frequency-domain and time-domain constraints for controlling the figures of merit, respectively. Combining these two techniques results in numerically efficient algorithms for designing optimized CMTs (or CMFBs) with a reduced computational complexity (number of arithmetic operations per output sample), particularly when both branches of the FRM structure are required. Design examples are included illustrating the efficiency of the design methods and the high performance of the resulting CMT structures.

Design of high-resolution cosine-modulated transmultiplexers with sharp transition band

Due to the growing importance of multichannel modulation, there has been great interest in the design of high-performance transmultiplex systems. In this paper, a new cosine-modulated transmultiplex structure is proposed based on a prototype filter designed with the frequency-response masking (FRM) approach. This new structure leads to substantial reduction in the computational complexity (number of multiplications per output sample) of the prototype filters having sharp transition band and equivalently small roll-off values. The relation between the interpolation factor used in the FRM prototype filter and the decimation factor in the subbands leads to distinct structures. Examples included indicate that the reduction in computational complexity can be higher than 50% of the current state-of-art designs, whereas the reduction on the number of distinct coefficients of the prototype filter can be reduced even further (over 75%). As a result, the proposed approach allows the design of very selective subfilters for transmultiplexes with a very large number of subchannels.

Numerically efficient optimal design of cosine-modulated filter banks with peak-constrained least-squares behavior

Several current applications related to signal compression and representation and high-speed transmission require very selective filter banks/transmultiplexers. A possible solution is to employ the cosine-modulated filter banks/transmultiplexers (CMFBTs) where the prototype filters satisfy demanding constraints with respect to both the total stopband energy and maximum stopband ripple. This work proposes an efficient procedure to design nearly-perfect reconstruction CMFBT prototype filters with peak-constrained least-squares characteristics using a modified weighted least-squares algorithm. Substantial flexibility is added in the design of the magnitude response of the prototype filter, ranging from minimum stopband energy to minimum stopband ripple, which may be required in many applications. Some constraints are imposed to the CMFBTs in order to control the direct transfer and aliasing distortion functions, related to the intercarrier and intersymbol interferences. Algebraic simplifications are also provided on the overall objective function and associated constraints, leading to substantial reduction on the computational burden of the optimization process. The procedure is proven to be very powerful in designing CMFBT systems satisfying multiple constraints as indicated by numerical examples.

Design of cosine-modulated filter bank prototype filters using the frequency-response masking approach

We use the frequency-response masking (FRM) approach to design prototype filters for cosine-modulated filter banks in the nearly perfect reconstruction case. With such an approach, it is possible to design a FRM filter with overall order almost equal to the direct-form FIR design, with only slight changes in the values of the inter-carrier and inter-symbol interferences and the attenuation of the bank filters. The result is an efficient design with reduced number of multipliers for the overall structure.

On the quality assessment of sound signals

This paper constitutes an introduction to the field of quality evaluation of sound (speech and audio) signals. The need for such an assessment is inherent to modern communications: VoIP, mobile phone, or teleconference systems require meaningful measures of performance, which may ultimately assure good service or profitable business. A brief survey on subjective and objective evaluation methods is provided. Recent developments as well as new topics to be investigated are also addressed. Experiments are conducted to illustrate how to validate quality assessment methods.

High-selectivity filter banks for spectral analysis of music signals

This paper approaches, under a unified framework, several algorithms for the spectral analysis of musical signals. Such algorithms include the fast Fourier transform (FFT), the fast filter bank (FFB), the constant- transform (CT), and the bounded- transform (BT), previously known from the associated literature. Two new methods are then introduced, namely, the constant- fast filter bank (CFFB) and the bounded- fast filter bank (BFFB), combining the positive characteristics of the previously mentioned algorithms. The provided analyses indicate that the proposed BFFB achieves an excellent compromise between the reduced computational effort of the FFT, the high selectivity of each output channel of the FFB, and the efficient distribution of frequency channels associated to the CT and BT methods. Examples are included to illustrate the performances of these methods in the spectral analysis of music signals.

ON WLS-CHEBYSHEV FIR DIGITAL FILTERS

A new numerical approach for designing FIR digital filters is proposed. The method is able to compromise maximum stopband attenuation and minimum stopband energy requirements. The approach is based on the weighted-least-squares (WLS) method using, at each iteration, a different weight function, which is made constant within a given frequency interval. In that manner, digital filters with partially equiripple and partially WLS-like stopbands are efficiently obtained. Generality of the method makes it suitable for the design of linear- and arbitrary-phase FIR filters.

A blind algorithm for reverberation-time estimation using subband decomposition of speech signals

An algorithm for blind estimation of reverberation time (RT) in speech signals is proposed. Analysis is restricted to the free-decaying regions of the signal, where the reverberation effect dominates, yielding a more accurate RT estimate at a reduced computational cost. A spectral decomposition is performed on the reverberant signal and partial RT estimates are determined in all signal subbands, providing more data to the statistical-analysis stage of the algorithm, which yields the final RT estimate. Algorithm performance is assessed using two distinct speech databases, achieving 91% and 97% correlation with the RTs measured by a standard nonblind method, indicating that the proposed method blindly estimates the RT in a reliable and consistent manner.

EFFICIENT DESIGN OF NARROWBAND COSINE-MODULATED FILTER BANKS USING A TWO-STAGE FREQUENCY-RESPONSE MASKING APPROACH

A new cosine-modulated filter bank (CMFB) structure is proposed based on the frequency-response masking (FRM) approach using masking filter decomposition. The resulting structure, the so-called FRM2-CMFB, presents reasonable computational complexity (number of arithmetic operations per output sample) and allows one to design filter banks with extremely large number of bands. The examples include the use of M=1024 bands, where the standard minimax method cannot be employed. These examples indicate that the reduction in computational complexity can be as high as 60% of the original FRM-CMFB structure, which does not use masking filter decomposition.