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Objective Electrically evoked auditory steady-state responses (EASSRs) are neural potentials measured in the electroencephalogram (EEG) in response to periodic pulse trains presented, for example, through a cochlear implant (CI). EASSRs could potentially be used for objective CI fitting. However, EEG signals are contaminated with electrical CI artifacts. In this paper, we characterized the CI artifacts for monopolar mode stimulation and evaluated at which pulse rate, linear interpolation over the signal part contaminated with CI artifact is successful. Methods CI artifacts were characterized by means of their amplitude growth functions and duration. Results CI artifact durations were between 0.7 and 1.7 ms, at contralateral recording electrodes. At ipsilateral recording electrodes, CI artifact durations are range from 0.7 to larger than 2 ms. Conclusion At contralateral recording electrodes, the artifact was shorter than the interpulse interval across subjects for 500 pps, which was not always the case for 900 pps. Significance CI artifact-free EASSRs are crucial for reliable CI fitting and neuroscience research. The CI artifact has been characterized and linear interpolation allows to remove it at contralateral recording electrodes for stimulation at 500 pps. [ABSTRACT FROM AUTHOR]

Abstract: This paper presents weighted approaches for integrated active noise control and noise reduction in hearing aids. The unweighted integrated active noise control and noise reduction scheme introduced in the previous work does not allow to trade-off between the active noise control and the noise reduction. In some circumstances it will, however, be useful to emphasize one of the functional blocks. Changing the original optimisation problem to a constrained optimisation problem leads to a scheme based on a weighted mean squared error criterion that allows to focus either on the active noise control or on the noise reduction. It is similarly possible to derive a scheme that allows to focus either on reducing the speech distortion or on reducing the residual noise at the eardrum. In a single speech source scenario and when the number of sound sources (speech plus noise sources) is less than or equal to the number of microphones, it is possible to derive a simple formula for the output signal-to-noise ratio of the latter scheme. It can then be shown that this scheme delivers a constant signal-to-noise ratio at the eardrum for any weighting factor. [Copyright &y& Elsevier]

Abstract: The Fiedler vector of a graph is the eigenvector corresponding to the smallest non-trivial eigenvalue of the graph''s Laplacian matrix. The entries of the Fiedler vector are known to provide a powerful heuristic for topology inference, e.g., to identify densely connected node clusters, to search for bottleneck links in the information dissemination, or to increase the overall connectivity of the network. In this paper, we consider ad hoc networks where the nodes can process and exchange data in a synchronous fashion, and we propose a distributed algorithm for in-network estimation of the Fiedler vector and the algebraic connectivity of the corresponding network graph. The algorithm is fully scalable with respect to the network size in terms of per-node computational complexity and data transmission. Simulation results demonstrate the performance of the algorithm. [Copyright &y& Elsevier]

Abstract: Per-tone pulse shaping has been proposed as an alternative to time domain spectral shaping for DMT transmitters. It shapes the spectrum of individual tones such that their stop band energy contribution can be minimized. In per-tone pulse shaping based DMT transmitters a fixed order non-sparse pulse shaping filter is typically used for every tone. Furthermore, no specific power loading scheme is used. However, the contribution of a particular tone to the stop band energy depends on the amount of power transmitted on the tone and the distance of the tone from the band edges. Tones with sufficiently low power as well as tones sufficiently away from the band edges could be assigned a sparse pulse shaping filter, which could then help to reduce the overall number of filter taps. In general the combination of power loading and sparse pulse shaping filters can be used to achieve a high data rate under given resource constraints. In this paper we present an algorithm to optimally allocate the available resources, i.e., power and filter taps, under PSD mask constraints, which is based on a dual problem formulation. This algorithm achieves a high performance with relatively low complexity, which is also demonstrated by simulations. [Copyright &y& Elsevier]

Abstract: In this paper, we consider distributed signal estimation in sensor networks where the nodes exchange compressed sensor signal observations to estimate different node-specific signals. In particular, we revisit the so-called distributed adaptive node-specific signal estimation (DANSE) algorithm, which applies to the case where the nodes share a so-called ‘common interest’, and cast it in the more general setting where the nodes have ‘different interests’. We prove existence of an equilibrium state for such a setting by using a result from fixed point theory. By establishing a link between the DANSE algorithm and game theory, we point out that any equilibrium of the DANSE algorithm is a Nash equilibrium of the corresponding game. This provides an intuitive interpretation to the resulting signal estimators. The equilibrium state existence proof also reveals a problem with discontinuities in the DANSE update function, which may result in non-convergence of the algorithm. However, since these discontinuities are identifiable, they can easily be avoided by applying a minor heuristic modification to the algorithm. We demonstrate the effectiveness of this modification by means of numerical examples. [Copyright &y& Elsevier]

Abstract: In current DSL systems, crosstalk is a major source of performance degradation. For upstream VDSL scenarios with in-domain crosstalk and AWGN, it has been shown that the crosstalk can be effectively mitigated using a linear zero-forcing canceler. Furthermore, the complexity can be reduced by only canceling the crosstalk from major crosstalkers, which is referred to as partial crosstalk cancellation. However, such approach does not work well in scenarios that also have out-of-domain (alien) crosstalk. As alien crosstalk is spatially correlated, the zero-forcing canceler performs very poorly, hence alternative structures for the canceler have to be examined. In this paper, we first demonstrate that an MMSE-based linear and nonlinear crosstalk canceler provides improved performance compared to the zero-forcing canceler in this scenario and then present efficient algorithms to perform partial crosstalk cancellation with these MMSE-based cancelers. [Copyright &y& Elsevier]

Abstract: The present study investigates the most important factors that limit the intelligibility of the cochlear implant (CI) processed speech in noisy environments. The electrical stimulation sequences provided in CIs are affected by the noise in the following three manners. First of all, the natural gaps in the speech are filled, which distorts the low-frequency ON/OFF modulations of the speech signal. Secondly, speech envelopes are distorted to include modulations of both speech and noise. Lastly, the N-of-M type of speech coding strategies may select the noise dominated channels instead of the dominant speech channels at low signal-to-noise ratio''s (SNRs). Different stimulation sequences are tested with CI subjects to study how these three noise effects individually limit the intelligibility of the CI processed speech. Tests are also conducted with normal hearing (NH) subjects using vocoded speech to identify any significant differences in the noise reduction requirements and speech distortion limitations between the two subject groups. Results indicate that compared to NH subjects CI subjects can tolerate significantly lower levels of steady state speech shaped noise in the speech gaps but at the same time can tolerate comparable levels of distortions in the speech segments. Furthermore, modulations in the stimulus current level have no effect on speech intelligibility as long as the channel selection remains ideal. Finally, wrong maxima selection together with the introduction of noise in the speech gaps significantly degrades the intelligibility. At low SNRs wrong maxima selection introduces interruptions in the speech and makes it difficult to fuse noisy and interrupted speech signals into a coherent speech stream. [Copyright &y& Elsevier]