the domain in which the signal parameters can be estimated unambiguously. This can be used in applications to avoid aliasing. Moreover, it has been shown that the mean squared error of PPS parameter estimates can be significantly decreased when a nonuniform sampling scheme is used [1]. Hence, from an accuracy and ambiguity point of view, a properly chosen nonuniform sampling scheme is preferred before a uniform scheme when estimating PPS parameters.Abstract-We present a robust noniterative algorithm to design optimal minimum-phase digital FIR filters with real or complex coefficients. We derive: 1) the discrete Hilbert transform (DHT) of the complex cepstrum of a causal complex minimum-phase sequence and 2) the minimum fast Fourier transform length for computing the DHT to achieve a desired coefficient accuracy.Abstract-Spectrally factorable Nyquist filters are used in data communications to avoid intersymbol interference. In this correspondence, an approach is developed for obtaining a Nyquist filter that is factorable having the smallest stopband energy for a given sidelobe level. The resulting constrained minimization problem is solved efficiently and reliably using the Goldfarb-Idnani algorithm. Some examples are presented comparing the present method with a previous approach from the literature. Index Terms-FIRNyquist filters, sidelobe level constraints. The authors are with the

We present the rst dual tone multiple frequency DTMF signal detector that meets the International Telecommunications Union ITU Q.24 DTMF standard when implemented on an 8-bit microcontroller. Key innovations include the use of adaptive notch lters and sophisticated decision logic. The DTMF detector is well suited for a multichannel digital signal processor implementation.

We present an adaptive enhancement method for speech signal corrupted by the 3 Tesla (3T) functional Magnetic Resonance Imaging (fMRI) acoustic noise. The noisy speech signal is filtered into several frequency bands and noise cancellation is applied to each band independently. We use Normalized Least Mean Squares (NLMS) algorithm for updating the co-efficients of the adaptive filters in both wideband and in multi-band implementations. The NLMS algorithm is used to ensure faster convergence and stability.The multi-band method shows a better performance than the conventional wideband method resulting in a better noise reduction. It is shown that the distortion introduced by the band-pass filters limits the overall speech enhancement when the number of bands increases beyond an optimal value.

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