A novel lattice-based adaptive infinite impulse response (IIR) notch filter is developed which features independent tuning of the notch frequency and attenuation bandwidth. The internal structure is based on planar rotators, ensuring reliable numerical behaviour and high processing rates in CORDIC environments. A simple update law allows a simpler implementation than previously proposed designs. Rather than minimizing an output error cost function, the algorithm is designed to achieve a stable associated differential equation, resulting in a globally convergent unbiased frequency estimator in the single sinusoid case, independent of the notch filter bandwidth. Using a second-order structure in the multiple sinusoid case, unbiased estimation of one of the input frequencies is achieved by thinning the notch bandwidth. The tracking behavior is superior to conventional output error designs, and the estimation of extremal frequencies is less prone to overflow instability than previously reported structures. >

An improved lattice-based adaptive IIR notch filter

A lattice notch filter for the detection and enchancement of sinusoids in noise is presented, and the convergence properties of the algorithm are investigated using ODE (ordinary differential equation) analysis. It is shown that the frequency estimate of the algorithm for a single sinusoid is unbiased, regardless of the pole-zero contraction factor and the noise variance. An expression for the MSE (mean square error) of the frequency estimate and the output SNR of the algorithm is derived. Simulation results are presented to verify the theoretical analysis. Based on cascading the proposed second-order adaptive notch filter, three different approaches for the detection of multiple sinusoids are presented. The first approach uses a linear cascade structure, and the second and third approaches use a triangular cascade structure. The performance of the algorithms for multiple sinusoids is investigated, and computer simulation results are presented and discussed. >

On the adaptive lattice notch filter for the detection of sinusoids

In this paper, we discuss the compression of waveforms obtained from measurements of power system quantities and analyze the reasons why its importance is growing with the advent of smart grid systems. While generation and transmission networks already use a considerable number of automation and measurement devices, a large number of smart monitors and meters are to be deployed in the distribution network to allow broad observability and real-time monitoring. This situation creates new requirements concerning the communication interface, computational intelligence and the ability to process data or signals and also to share information. Therefore, a considerable increase in data exchange and in storage is likely to occur. In this context, one must achieve an efficient use of channel communication bandwidth and a reduced need of storage space for power system data. Here, we review the main compression techniques devised for electric signal waveforms providing an overview of the achievements obtained in the past decades. Additionally, we envision some smart grid scenarios emphasizing open research issues regarding compression of electric signal waveforms. We expect that this paper will contribute to motivate joint research efforts between electrical power system and signal processing communities in the area of signal waveform compression.

The Compression of Electric Signal Waveforms for Smart Grids: State of the Art and Future Trends

Feedback cancellation in hearing aids involves estimating the feedback signal and subtracting it from the microphone input signal. The feedback-cancellation system described updates the estimated feedback path whenever changes are detected in the feedback behavior. When a change is detected, the normal hearing-aid processing is interrupted, a pseudorandom probe signal is injected into the system, and a set of filter coefficients is adjusted to give an estimate of the feedback path. The hearing aid is then returned to normal operation with the feedback-cancellation filter as part of the system. Two approaches are investigated for computing the filter coefficients: a least-mean square (LMS) adaptive filter and a Wiener filter. Test results are presented for a computer simulation of an in-the-ear (ITE) hearing aid. The simulation results indicate that more than 10 dB of cancellation can be obtained and that the Wiener filter is more effective in the presence of strong interference. >

Feedback cancellation in hearing aids: results from a computer simulation

A new technique for IIR multiple notch filter design is proposed. The specification of notch filter is first transformed into that of allpass filter. Then, we develop an effective approach to design this desired allpass filter. The realization of proposed notch filter is equivalent to the realization of an allpass filter. Due to the mirror-image symmetry relation between the numerator and denominator polynomials of allpass filter, the notch filter can be realized by a computationally efficient lattice structure with very low sensitivity. Moreover, some examples are presented to examine the effectiveness of proposed method.

/pdf/iir-multiple-notch-filter-design-based-on-allpass-filter-4kgw02c5ej.pdf

IIR multiple notch filter design based on allpass filter

An adaptive notch filter based on a fast recursive-least-squares (FLS) algorithm is introduced. The structure is canonical and consists of a transversal section cascaded with an IIR section. The adaptive procedure takes place in the transversal section, and the impact of the notch factor is pointed out. Overall, the FLS notch filter has about the same complexity as the fast transversal adaptive filter, but it performs better for the analysis of narrowband signals in high-level noise. >

Fast least squares adaptive notch filtering

An adaptive notch filter based on a fast recursive least squares (FLS) algorithm is studied. The adaptive procedure takes place in the FIR (finite impulse response) section of the filter and the IIR (infinite impulse response) section is updated each time with the most recent coefficients of the other section. The implementation can take the form of a cascade of second order sections. Overall, the FLS notch filter has about the same complexity as the transversal adaptive filter, but performs better for the analysis of narrow-band signals in high-level noise. >

J.M. Travassos-Romano

Papers

Fast least squares adaptive notch filtering

Fast least squares adaptive notch filtering