RMS delay and coherence bandwidth measurements in indoor radiochannels in the UHF band

A study of time dispersion in different indoor line-of-sight radio channels in the 492-862 MHz band is presented in this paper. A combined method to filter the noise in the measured impulse response is described. The effect of frequency windowing on the impulse responses and the root mean square (rms) delay spread is also investigated. It has been found that, in general, the use of windows with lower side-lobe levels yields larger values of the rms delay spread. The relation between the mean delay and the rms delay spread has also been studied for copolar and crosspolar channels. The dependence of the coherence bandwidth on the rms delay spread has been considered, and an inverse relation has been tested for both components     

A Novel Simple Semi-Empirical Model for the Effective Earth Radius Factor

This paper presents a dual cumulative probability distribution model for predicting the cumulative distribution of the probability that a particular value of the propagation k-factor will be exceeded. The distributions relate to so-called "dry" and "wet" (or "climatic") components. The model is based on meteorological radiosonde data observed over a period of some 11 years and published by other workers. The model is extended by a simple height regression analysis in order to predict the cumulative behavior of the k-factor at different ground level heights in the summer rainfall area of South Africa during different months. Comparisons are made between predicted and observed data. The model may have potential applications in other parts of the world where data on k-factor behavior are scarce     

A time domain approach for the fluctuation analysis of heart raterelated to instantaneous lung volume

A time domain technique for estimating transfer characteristics from fluctuations of instantaneous lung volume (ILV) to heart rate (HR) is presented. An effective procedure for estimating the impulse response of HR to ILV is proposed. Pre- and post-processing procedures, including prefiltering of the HR signal, preenhancement of the high frequency content of the ILV signal, and post-filtering of the estimated impulse response, together with a random breathing technique, are shown to effectively reduce spurious transfer gain so as to get a stable estimate of the impulse response. Analysis of the data collected from fourteen healthy male subjects in various conditions revealed that there are three components in the impulse response: fast positive, delayed slow negative, and oscillatory. The effects of the autonomic blocking agents propranolol and atropine on these transfer characteristics are also described     

Restoring Coefficient Symmetry in Polyphase Implementation of Linear-Phase FIR Filters

Polyphase implementation of FIR filters effectively reduces the multiplication rate and data storage in a multirate system. However, the coefficients of the polyphase components are no longer symmetric even though the overall filter has a symmetric (or anti-symmetric) impulse response. In this paper, we introduce a new technique that recasts pairs of the original polyphase components as sums or differences of auxiliary pairs of symmetric and anti-symmetric impulse response filters. The coefficient symmetry of these auxiliary polyphase components can be fully exploited to reduce arithmetic complexity without undue complications. Our new technique makes use of the fact that the impulse responses of the non-symmetric polyphase components exist in time-reversed pairs which can be synthesized from pairs of symmetric and anti-symmetric impulse response filters. This results in a factor-of-two reduction in the number of multipliers required to implement the polyphase components.     

Deconvolution of the impulse response of a conducting sphere by the conjugate gradient method

An experimental technique for determining the impulse response of a conducting sphere has been presented. A method of exciting and recording transient waveforms scattered from conducting spheres of various sizes has been described and extensively tested. A modified conjugate-gradient method has been applied to the transient waveforms to determine the impulse response of the objects. It is shown that by proper weighting in computing the residual in the conjugate-gradient method the oscillation due to noise could be reduced. A recently developed data-processing technique has been applied to these impulse responses to obtain their spectra, which yield adequately the resonant frequencies of the objects.     

Photonic wideband array antennas

Presents an introduction to the optical control of array antennas by using fiber optic links for remote control and a photonic time shift network for wide instantaneous bandwidth. An overview of the development of a wideband conformal array designed for airborne surveillance radars is given. The paper covers the system design and the performance of an L-band (850-1400 MHz) M-element array controlled by photonics. Packaging techniques of the photonic components and the array aperture are discussed. The wideband performance of the system is highlighted. A nano-second impulse response has been measured to demonstrate a 50% instantaneous bandwidth (550 MHz, 30 cm range resolution) for target ID and imaging. A built-in signal injection technique based on time-domain impulse measurement was used to calibrate the wideband components in the time-shift beamforming network     

Impulse response of sinc/sup N/ FIR filters

A simple recurrence formula is presented for computing the impulse response coefficients of the sinc/sup N/ FIR filter, consisting of a cascade of N sinc filters, each of length M. A closed form expression is also given for the first M coefficients.     

Least square identification of alias components of linear periodically time-varying systems and optimal training signal design

A least-squares (LS) method for identifying alias components of discrete linear periodically time-varying (LPTV) systems is proposed.The authors apply a periodic input signal to a finite impulse response (FIR)--LPTV system and measure the noise-contaminated output.The output of this LPTV system has the same period as the input when the period of the input signal is amultiple of the period of the LPTV system.The authors show that the input and the output can be related by using the discrete Fourier transform. In the frequency domain, an LS method can be used to identify the alias components. A lower bound on the mean square error (MSE) of the estimated alias components is given for FIR--LPTV systems.The optimal training signal achieving this lower MSE bound is designed subsequently. The algorithm is extended to the identification of infinite impulse response (IIR)--LPTV systems as well. Simulation results show the accuracy of the estimation and the efficiency of the optimal training signal design.     

Rapid identification of a sparse impulse response using an adaptive algorithm in the Haar domain

This paper proposes a fast convergence adaptive algorithm for identifying a sparse impulse response that is rich in spectral content. A sparse impulse response is referred here as a discrete time impulse response that has a large number of zero or near zero coefficients. The basic idea for rapid identification is to automatically determine the locations of the nonzero impulse response coefficients for their adaptation and eliminate the unnecessary adaptation of zero coefficients. The proposed method, which is called the Haar-Basis algorithm, employs a transform approach by modeling the sparse impulse response in the Haar domain. The Haar transform has many basis sets and each of them contains basis vectors that span the entire time domain range. This special nature of the Haar transform allows for the selection of one small subset of adaptive filter coefficients whose basis vectors span the entire range of the impulse response. These coefficients are adapted at the beginning and are then used subsequently to identify, from the hierarchical structure of the Haar transform, the rest of the filter coefficients that must be adapted to correctly model the unknown sparse impulse response. The consequence is avoiding adaptation of many zero coefficients, leading to a dramatic improvement in either convergence speed or steady state excess mean-square error (EASE), while requiring no a priori knowledge such as the number of nonzero coefficients in the unknown sparse impulse response. The proposed algorithm has been tested with a variety of unknown sparse systems using both white noise input and colored input whose spectrum closely resembles that of speech. Simulation results show that the new approach provides promising results.     

Digital filter design techniques in the frequency domain

Digital filtering is the process of spectrum shaping using digital components as the basic elements. Increasing speed and decreasing size and cost of digital components make it likely that digital filtering, already used extensively in the computer simulation of analog filters, will perform, in real-time devices, the functions which are now performed almost exclusively by analog components. In this paper, using the z-transform calculus, several digital filter design techniques are reviewed, and new ones are presented. One technique can be used to design a digital filter whose impulse response is like that of a given analog filter; other techniques are suitable for the design of a digital filter meeting frequency response criteria. Another technique yields digital filters with linear phase, specified frequency response, and controlled impulse response duration. The effect of digital arithmetic on the behavior of digital filters is also considered.     

Removal of Base-Line Wander and Power-Line Interference from the ECG by an Efficient FIR Filter with a Reduced Number of Taps

Linear phase filtering is proposed for the removal of baseline wander and power-line frequency components in electrocardiograms. In order to reduce the large number of computations involved in the digital filtering that are necessary, the desired filter spectrum was defined periodically. Making use of the property that the spectrum period is 50 Hz, the spectrum can be realized with a considerably reduced number of impulse response coefficients. This, in combination with the necessary impulse response symmetry, leads to a reduction in the number of multiplications per output sample by a factor of 10. A suitable impulse response is designed with a pass-band ripple of less than 0.5 dB and a high stop-band attenuation. The applicability is demonstrated by applying the filtering to exercise electrocardiograms.     

Statistical analysis of measured impulse response functions of 2.0GHz indoor radio channels

The measurements of the impulse response of a 2.0 GHz indoor radio channel are reported. A statistical analysis of the characteristics of the amplitude of multipath components is presented. In particular, the spatial correlation of the single multipath components and the cross-correlation between the amplitudes of adjacent multipath components have been determined. Conclusions are drawn with regard to the adequacy of the wide sense uncorrelated scattering model as a consistent model for the indoor radio channel     

Impulse response modeling of indoor radio propagation channels

If indoor radio propagation channels are modeled as linear filters, they can be characterized by reporting the parameters of their equivalent impulse response functions. The measurement and modeling of estimates for such functions in two different office buildings are reported. The resulting data base consists of 12000 impulse response estimates of the channel that were obtained by inverse Fourier transforming of the channel's transfer functions. It is shown that the number of multipath components in each impulse response estimate is a normally-distributed random variable with a mean value that increases with increasing antenna separations; a modified Poisson distribution shows a good fit to the arrival time of the multipath components; amplitudes are lognormally distributed over both local and global areas, with a log-mean value that decreases almost linearly with increasing excess delay; for small displacements of the receiving antenna, the amplitude of the multipath components are correlated; the amplitudes of adjacent multipath components of the same impulse response function show negligible correlations; and the RMS delay spread over large areas is normally distributed with mean values that increase with increasing antenna separation     

The application of two-dimensional signal transformations to theanalysis and synthesis of structural excitations observed in acousticalscattering

Acoustic scattering from air-filled, elastic shells submerged in water is an important problem in applied science. The excitations of interest yield a set of physically distinct components to the impulse response of a shell. The components form a natural basis for all signals which can be observed in acoustical scattering experiments from the shell via temporal convolution with some chosen input signal. The Fourier transform (FT) of the impulse response of a shell yields its transfer function, which is also called the form function. We study two types of shells in this paper: a spherical shell, and a finite, ribbed, cylindrical shell with endcaps. Utilizing several different two-dimensional (2-D) signal transformations, we can decompose the response of the shells. The resulting 2-D images allow for a striking visual decomposition of the responses into their distinct components. In the case of the spherical shell, a virtually exact theory exists that allows for analytic synthesis of the shell response into its components. However, for the more complex cylindrical shell, the theory for the direct scattering problem is not nearly so mature. Yet, we can still decompose experimentally-obtained shell responses into their distinct components via signal synthesis techniques applied to the 2-D transforms     

Almost Equiripple Low-Pass FIR Filters

An approximation of the linear phase almost equiripple low-pass finite impulse response filter is introduced. The frequency response of an almost equiripple low-pass finite impulse response filter closely approaches the frequency response of an optimal equiripple low-pass finite impulse response filter in the Chebyshev sense. The presented approximation is based on the generating polynomial. Despite that the generating polynomial has no iso-extremal behavior, it is related to the class of iso-extremal polynomials. The zero phase transfer function of an almost equiripple low-pass finite impulse response filter follows from the generating polynomial. The closed form solution for the direct algebraic computation of the impulse response of the filter has been developed on the basis of generalization of the differential equation suitable for the half-band specifications. No numerical procedures are involved. The practical design procedure based on the developed approximation is presented. For illustration of the design procedure one example of the design is included here.     

Logarithmic wavelength demultiplexers

A general approach for a full 1 /spl times/ N demultiplexer using a tree of filter stages is proposed. The device architecture is compact, requiring at least N - 1 filter stages, and flexible, as each filter stage can be arbitrarily designed, with the only constraint of the half-band power property. The filters can be realized using any optical filtering techniques as thin-film interference, Bragg gratings, or planar delay-line circuits. The performances of the proposed architecture are illustrated with respect to different lattice-form finite impulse response (FIR) and infinite impulse response (IIR) filter stages, showing that the demultiplexer inherently presents low crosstalk and flat passband. A design example of a 1 /spl times/ 4 demultiplexer consisting of three all-pass (AP) filters is compared with a generalized Mach-Zehnder interferometer (MZI) with four AP filters in its arms, showing that the two approaches achieve similar results.     

An adaptive technique for determining a reduced model for a system

A reduced or sparse system model is discussed that will contain only the most significant components, as opposed to a complete finite impulse response (FIR) model which may not be very accurate with the requirement of only a few components. The technique presented uses an adaptive delay filter to provide the sparse model and compares it to the model obtained with the standard adaptive filter     

Residual echo signal in critically sampled subband acoustic echocancellers based on IIR and FIR filter banks

The residual echo signal characteristics of critically sampled subband acoustic echo cancellers are analyzed. For finite impulse response (FIR) filter banks, the residual echo signal usually has a relatively broad spectral nature around the subband edges. The residual echo signal of power symmetric infinite impulse response (PS-IIR) filter banks, on the other hand, has very narrowband spectral components around the subband edges. These components can be efficiently removed with PS-IIR notch filters that integrate neatly into the filter banks without introducing perceptually noticeable degradation to the near-end speech. This solution has very low computational complexity and does not impinge on the system performance. Simulation studies with recordings from the cockpit of a car, based on a fast QR least-squares adaptive algorithm, demonstrate the potential of this approach for a practical AEC system     

Wide-band systems and Gaussianity (Corresp.)

The Berry-Esseén theorem has been used to show that narrow-band filtering tends to make non-Gaussian noise more Gaussian. In this correspondence, the same theorem is invoked to demonstrate that wide-band filtering can also make noise more Gaussian, provided the filter impulse response has a large time-bandwidth product.