Academy for Super Smart Society

As a fundamental building block of multirate systems, the downsampler, also known as the decimator, is a periodically time-varying linear system. An eigensignal of the downsampler is defined to be an input signal which appears at the output unaltered or scaled by a non-zero coefficient. In this paper, the eigensignals are studied and characterized in the time and z domains. The time-domain characterization is carried out using number theoretic principles, while the one-sided z-transform and Lambertform series are used for the transform-domain characterization. Examples of non-trivial eigensignals are provided. These include the special classes of multiplicative and completely multiplicative eigensignals. Moreover, the locus of poles of eigensignals with rational z transforms are identified.

A modified delta encoding method a to speech signal are proposed. In this m algorithm is applied to find the minimum dis two frames in speech signal and minimum spa to find an effective delta encoding path. I method is applied to the compression of sinu results show that the data size after compres than a usual delta encoding whose path permutated. In addition, the proposed method to apply on data security for practical use encoding path which can be used as the securit and long enough.

A new lattice structure is described. It is capable of implementing all paraunitary two-channel filter banks where the filters have complex coefficients and yield symmetric wavelet bases. This lattice structure, while being a general design method, can also be used to actually design the filter bank. These filter banks are, in fact, a special case of multifilter banks and can also be related to Golay-Rudin-Shapiro complementary polynomial pairs. The applications of such filter banks are to be found in subband coding and ...

An efficient approach to mitigate the howling effect in hearing aids is via the use of Acoustic Feedback Cancellation (AFC). In this paper, the use of a Forward Linear Predictor (FLP) is investigated to improve the performance of an AFC system in multi-band hearing aids. The FLP is used to predict the speech input signal before eliminating it from the error signal of the AFC system. Computer simulations demonstrate that more accurate estimation of the acoustic feedback signal than the conventional AFC system can be obtained. In addition, maximum usable gain of hearing aids required by the users can be achieved when employing the proposed multi-band AFC system.

The response of lowpass maximally at digital 'Iters to polyno-mial signals is analyzed by means of a formal algebraic method in the z domain. The general class of maximally at FIR 'Iters, en-compassing linear-phase, non-linear-phase, and half-band 'lters, is studied. A ...

Background: A new framework for heart sound analysis is proposed. One of the most difficult processes in heart sound analysis is segmentation, due to interference form murmurs. Method: Equal number of cardiac cycles were extracted from heart sounds with different heart rates using information from envelopes of autocorrelation functions without the need to label individual fundamental heart sounds (FHS). The complete method consists of envelope detection, calculation of cardiac cycle lengths using auto-correlation of envelope signals, features extraction using discrete wavelet transform, principal component analysis, and classification using neural network bagging predictors. Result: The proposed method was tested on a set of heart sounds obtained from several on-line databases and recorded with an electronic stethoscope. Geometric mean was used as performance index. Average classification performance using tenfold cross-validation was 0.92 for noise free case, 0.90 under white noise with 10 dB signal-to-noise ratio (SNR), and 0.90 under impulse noise up to 0.3 s duration. Conclusion: The proposed method showed promising results and high noise robustness to a wide range of heart sounds. However, more tests are needed to address any bias that may have been introduced by different sources of heart sounds in the current training set, and to concretely validate the method. Further work include building a new training set recorded from actual patients, then further evaluate the method based on this new training set.

Summary Learning management systems (LMS) offer the possibility of doing exams. These can be of different kinds, like multiple choices, large explanation and so on, using IMS QTI standard. Also a LMS has its permissions and restrictions associated. These permissions ...

Wave digital filters (WDF'S) are well known to have a low-sensitivity property in general. design approaches, new circuit realizations of series and parallel sections are proposed in this paper. Also, equivalent transformations are presented which enable new WDF's to be derived from those of Fettweis. Accordingly, many kinds of equivalent WDF's can be designed from a single analog prototype filter. The comparison of sensitivity characteristics of these circuits shows that far less sensitive WDF's can be realized by an appropriate choice from the conventional and the new sections depending on the value of their internal multiplier coefficients. In addition to several existing J 1-d G (z-') 1 l-G(z-') (Id) G (z-l)

When we realize a digital filter, a low-sensitivity structure is desired to produce less degradation of characteristics in finite word length. In addition, for the fast operation, a structure that can be parallelized or pipelined is preferable. The low-sensitivity digital filter, a wave digital filter proposed by Fettweis, is well known. Taking into consideration the forementioned, we propose synthesis of a digital filter that will permit pipeline processing and possess low-sensitivity as in a wave digital filter. First we show that a circuit structure using T-cascade connection of basic sections allows pipeline processing; we then derive a method for synthesis of a digital filter having this form of structure from a given transfer function in z-domain. Then in the proposed method we demonstrate synthesis of not only IIR filters but also FIR filters in a unified fashion. A realization of basic sections used in circuit synthesis is shown and the effectiveness of our synthesis method is verified by examples.

A novel multiplierless structure for maximally flat linear phase FIR digital filters is proposed. The structure is based on a recursive evaluation technique for Bernstein polynomials. It consists of identical multiplierless building blocks connected in a regular fashion.

This paper proposes a novel and robust voice activity detection (VAD) algorithm utilizing long-term spectral flatness measure (LSFM) which is capable of working at 10 dB and lower signal-to-noise ratios(SNRs). This new LSFM-based VAD improves speech detection robustness in various noisy environments by employing a low-variance spectrum estimate and an adaptive threshold. The discriminative power of the new LSFM feature is shown by conducting an analysis of the speech/non-speech LSFM distributions. The proposed algorithm was evaluated under 12 types of noises (11 from NOISEX-92 and speech-shaped noise) and five types of SNR in core TIMIT test corpus. Comparisons with three modern standardized algorithms (ETSI adaptive multi-rate (AMR) options AMR1 and AMR2 and ITU-T G.729) demonstrate that our proposed LSFM-based VAD scheme achieved the best average accuracy rate. A long-term signal variability (LTSV)-based VAD scheme is also compared with our proposed method. The results show that our proposed algorithm outperforms the LTSV-based VAD scheme for most of the noises considered including difficult noises like machine gun noise and speech babble noise.

Two-dimensional (2-D) linear-phase diamond-shaped (DS) FIR digital filters have important image processing applications. They are used in sampling structure conversion, e.g. interlace-to-progressive conversion. Maximally flat (MF) DS filters have attractive time-and frequency-domain properties and are preferred in some applications. In this correspondence MF half-band DS FIR are designed using the twovariable Bernstein polynomial. Formulas for the transfer function and the impulse response are obtained.

New adaptive algorithms based on multi-band decomposition of the error signal and application of a different convergence factor for each band are derived. With this approach, tracking ability and performance in steady state can be traded off along the frequency domain giving rise to estimates of the adaptive filter coefficients closer to the ideal response as compared to those obtained with conventional least-mean-square (LMS) and recursive least-squares (RLS) algorithms, particularly when the statistical properties of the analyzed signal vary along the frequency spectrum. A new adaptation technique for the forgetting factor depending exclusively on the autocorrelation values of the input signal is also introduced and a multi-band RLS algorithm, with an independent variable forgetting factor for each band, suitable for the analysis of nonstationary signals is described. Computer experiments comparing the performance of multi-band and conventional LMS and RLS algorithms are shown.

Combining the methods of generating functions and discrete-time linear systems, we can develop multidimensional (M-D) digital systems for systematic generation of entire families of interrelated digital filters. These M-D systems, dubbed filter-generating systems, are mostly of infinite-impulse response (IIR) type and can produce the impulse response sequence of any member of the family. Temporal and spatio-temporal implementation of filter-generating systems are studied. A temporal implementation scheme is devised for causal filter-generating systems in order to implement an arbitrary filter member using (M-D) signal processing techniques. It is shown that this may result in low order recursive filtering structures that can implement arbitrarily high order members of the family. A spatio-temporal implementation scheme is developed for generation of regular signal flow-graphs for the generated family of finite-impulse response (FIR) filters. It is shown that the signal flow-graphs give rise to cellular array structures. Concrete examples are provided for maximally flat FIR filters.

Coiflets are filter banks, where the sum of the number of vanishing moments of the analysis and synthesis limit functions is maximum for a given support width. It is known how to design biorthogonal coiflets with odd-length filters. However, the precise relationship among the vanishing moments of the analysis and synthesis scaling functions is unknown. This is the first problem solved in this correspondence. Second, biorthogonal coiflets with even-length filters, which have remained unknown previously, are designed and the relationship among the vanishing moments of the analysis and synthesis limit functions shown. A generalization of the Bernstein polynomial is advanced. Each of these two wavelet families is parametrized by three integers. The design is based on explicit formulae.

A motion blur identification scheme is proposed for nonlinear uniform motion blurs approximated by piecewise linear models which consist of more than one linear motion component. The proposed scheme includes three modules that are a motion direction estimator, a motion length estimator and a motion combination selector. In order to identify the motion directions, the proposed scheme is based on a trial restoration by using directional forward ramp motion blurs along different directions and an analysis of directional information via frequency domain by using a Radon transform. Autocorrelation functions of image derivatives along several directions are employed for estimation of the motion lengths. A proper motion combination is identified by analyzing local autocorrelation functions of non-flat component of trial restored results. Experimental examples of simulated and real world blurred images are given to demonstrate a promising performance of the proposed scheme.