Low-pass Signal Research Articles

A universal interpolative filter for low-pass and bandpass signals—CSINC interpolator

A FIR csinc interpolative filter design for bandpass signals is presented in this paper. The filter coefficients are calculated from the covariance of basis functions for the bandpass function space. They are invariant with respect to the samples of the input signals. The csinc model also works for low-pass signals as one of the design parameters approaches zero. The performance of the csinc filter is analyzed from the perspectives of accuracy, stability, and adaptation to noise in the samples. Comparisons are made between the csinc model and the existing interpolation models in the literature.

Digital time-delay beamforming with interpolated signals

In this paper, a general treatment of digital time-delay beamforming with interpolated signals is given. It covers beamforming for low-pass and band-pass signals, and popular digital beamformer structures are included in this approach. Two different topics are combined in the investigation: the first is reconstruction of complex envelope and analytic band-pass signals from a finite number of samples, the second topic applies the first to digital beamforming. Closed form expressions for the digital interpolation beamformer output and its beam-pattern are derived, and the effect of sampling and interpolation is discussed. Numerical simulations underline the power of the general approach of choosing the right sampling rate and interpolation method to realize digital beamformers with very little distortion of their beam-pattern.

Voltage-mode highpass, bandpass and lowpass filters using two DDCCs

A new configuration for realizing voltage-mode highpass, bandpass and lowpass filters, simultaneously, using two differential difference current conveyors (DDCCs), two grounded capacitors and two grounded resistors is presented. The proposed circuit offers the following features: realization of highpass, bandpass and lowpass signals from the same configuration, no requirements for component matching conditions, the use of only grounded capacitors and resistors, high input impedance, low active and passive sensitivities and simpler configuration due to the use of only two DDCCs and four passive components.

Mean Frequency Estimation of Narrowband Signals

This letter shows that the single frequency approximation for a narrowband lowpass signal embedded in white noise using the Pisarenko harmonic decomposition algorithm is approximately the power-weighted mean frequency of the signal. Experimental results indicate that this method is superior to a commonly used Fourier transform based mean frequency estimation method.

An empirical study of minimax-optimal fractional delays for low-pass signals

We consider the design of fractional delay (FD) filters for low-pass signals using the minimax-optimality criterion. In particular, we present an empirically derived relationship between the bandwidth, filter order, delay, and peak error, which is useful for parameter selection in design problems. We also present a simple method for rapid online calculation of FD filters with arbitrary shifts. Finally, we present some numerical results comparing these minimax-optimal FD filters with filters derived via the generalized least squares, Lagrange interpolation, and approximate prolate series windowed methods. The simulations suggest that the minimax-optimal delays are competitive with filters designed by other means, and in some cases can significantly outperform them.

A multimedia wireless transmission system using complex‐coefficient multirate filter banks

AbstractA method of multimedia transmission in wireless communication is to apply the frequency‐division multiplex scheme, in which the bandwidth is assigned flexibly according to the signal transmission rate. The purpose of this paper is to develop a system in which multimedia wireless transmission is handled efficiently in the downlink, and to propose a method based on the complex‐coefficient multirate filter bank. The complex‐coefficient multirate filter bank is a technique which is suited to signal processing in the equivalent low‐pass signal system in wireless communication. In the proposed method, bandwidth can be assigned flexibly to multiple signals with various transmission rates, according to their rates. An analysis of the proposed scheme shows that the proposed system is suited to multimedia wireless transmission. © 2002 Scripta Technica, Electron Comm Jpn Pt 2, 85(3): 58–67, 2002

Potential use of the undersampling technique in the acquisition of nuclear magnetic resonance signals

This communication reviews the use of undersampling techniques to acquire NMR signals. Undersampling transforms bandpass free induction decay (FID) signals, centered at high frequencies, into lowpass signals or bandpass signals centered at much lower frequencies. Consequently, the analog electronic stages that perform the demodulation can be eliminated, gaining in stability and reducing the phase distortion while maintaining an equivalent or better signal to noise ratio when an adequate sampling rate is chosen. The technique has been tested on a BRUKER BIOSPEC BMT 47/40, and the results show that undersampling could be used to process NMR and MRI signals, extending the range of applications of the ‘digital radio’ techniques to NMR and MRI apparatus.

Potential use of the undersampling technique in the acquisition of nuclear magnetic resonance signals

This communication reviews the use of undersampling techniques to acquire NMR signals. Undersampling transforms bandpass free induction decay (FID) signals, centered at high frequencies, into lowpass signals or bandpass signals centered at much lower frequencies. Consequently, the analog electronic stages that perform the demodulation can be eliminated, gaining in stability and reducing the phase distortion while maintaining an equivalent or better signal to noise ratio when an adequate sampling rate is chosen. The technique has been tested on a BRUKER BIOSPEC BMT 47/40, and the results show that undersampling could be used to process NMR and MRI signals, extending the range of applications of the 'digital radio' techniques to NMR and MRI apparatus.

Voltage-mode multifunction filter using one current feedback amplifier and one voltage follower

A universal voltage-mode second-order filter with three inputs and one output employing one current feedback amplifier, one voltage follower, two capacitors and two resistors is presented. It offers the following features: realization of allpass, notch, highpass, bandpass and lowpass signals from the same configuration, no requirements for component-matching conditions, orthogonal control of natural frequency and bandwidth, low active and passive sensitivities, minimum active and passive components and cascadability.

Higher order sampling and recovering of lowpass signals

A higher order sampling function applied to lowpass signals is described. To recover the original signal from its periodic nonuniform samples, a formulation of the sampling function is introduced, based on the occurrence of local gaps in the sampled signal spectrum. As a result, the interpolant is greatly simplified.

KALP: a Kalman filter-based adaptive clock method with low-pass prefiltering for packet networks use

We consider the issue of source clock frequency recovery in packet networks and propose a new adaptive clock method based on the Kalman filter (KF) with low-pass prefiltering-KALP (Kalman filter-based Adaptive clock method with Low-pass Prefiltering) in short. Noting that because of the difficulty in modeling as well as the nonwhite Gaussian nature of the packet jitter most existing adaptive clock methods could not successfully adopt the Kalman filter, we take a new approach to packet jitter modeling for the KALP. We model the packet jitter not directly but after shaping its characteristics by low-pass prefiltering. This low-pass prefiltering is an important arrangement as it helps to convert the packet jitter into a low-pass signal regardless of its original characteristics, thus enabling to model the prefiltered packet jitter using a simple first-order autoregressive [AR(1)] process. The low-pass prefilter should be carefully selected not to lose the timing information while prefiltering, and the moving averager employed in this paper satisfies this requirement. The AR(1)-modeled jitter component is amenable to the KF-based processing, which in this case becomes an optimal estimator. The design parameters including the initial conditions of the KF and AR(1) parameters can be determined based on the service clock specification and packet interarrival times during the delay smoothing process. We carry out various simulations to compare the performance of the KALP with the existing buffer-based adaptive clock method and demonstrate that the KALP can significantly reduce the fluctuation in the level of receiving buffer as well as the time to recover the source clock frequency.

Harmonic analysis of spiking neuronal pairs

Harmonic analysis is applied to analyze the transmission of bandlimited signals via spike trains generated by a pair of leaky integrate-and-fire (LIF) model neurons organized in a push–pull, or ‘on/off’, configuration. It is found that lowpass signals can be transmitted with only moderate distortion utilizing mean spike rates with many spikes per signal coherence time. The transmission of bandpass signals is accomplished through phase locking of the output spikes to the input signal over a wide range of frequencies, using one or two spikes per coherence time of the signal, but in this case with limited dynamic range in amplitude. Noise-free LIF-generated spike trains decoded with linear filters can transmit information at an average rate of 2–3 bits per spike.

Primate pleuroesophageal tissue barrier frequency response and esophageal pressure waveform bandwidth in health and acute lung injury.

Dynamic intraesophageal pressure (Pes) is used to estimate intrapleural pressure (Ppl) to calculate lung compliance and resistance. This study investigated the nonhuman primate Ppl-Pes tissue barrier frequency response and the dynamic response requirements of Pes manometers. In healthy monkeys and monkeys with acute lung injury undergoing ventilation, simultaneous Ppl and Pes were measured directly to determine the Ppl-Pes tissue barrier amplitude frequency response, using the swept-sine wave technique. The bandwidths of physiologic Pes waveforms acquired during conventional mechanical ventilation were calculated using digital low-pass signal filtering. The Ppl-Pes tissue barrier is amplitude-uniform within the bandwidth of conventional Pes waveforms in healthy and acute lung injury lungs, and does not significantly attenuate Ppl-Pes signal transmission between 1 and 40 Hz. At Pes frequencies higher than conventional clinical regions of interest the Ppl-Pes barrier resonates significantly, is pressure amplitude dependent at low-pressure offsets, and is significantly altered by acute lung injury. Allowing for 5% or less Pes waveform error, the maximum Pes bandwidths during conventional ventilation were 1.9 Hz and 3.4 Hz for physiologic and extreme-case waveforms in healthy lungs and 4.6 Hz and 8.5 Hz during acute lung injury. In monkeys, the Ppl-Pes tissue barrier has a frequency response suitable for Ppl estimation during low-frequency mechanical ventilation, and Pes manometers should have a minimum uniform frequency response up to 8.5 Hz. However, the Ppl-Pes tissue barrier adversely affects the accurate estimation of dynamic Ppl at high frequencies, with varied airway pressure amplitudes and offsets, such as the Ppl encountered during high-frequency oscillatory ventilation.

Optical subband coding system

We propose an optical subband coding scheme, which consists of subband decomposition and reconstruction using multichannel spatial filtering. The undersampled subband signals may be considered as a multiresolution representation. A passband signal can be converted down to a lowpass signal through a combination of undersampling and lowpass filtering. Conversely, a lowpass signal can be shifted up to a bandpass signal using a combination of oversampling and bandpass filtering.

Nonexpansive pyramid for image coding using a nonlinear filterbank

A nonexpansive pyramidal decomposition is proposed for low-complexity image coding. The image is decomposed through a nonlinear filterbank into low- and highpass signals and the recursion of the filterbank over the lowpass signal generates a pyramid resembling that of the octave wavelet transform. The structure itself guarantees perfect reconstruction and we have chosen nonlinear filters for performance reasons. The transformed samples are grouped into square blocks and used to replace the discrete cosine transform (DCT) in the Joint Photographic Expert Group (JPEG) coder. The proposed coder has some advantages over the DCT-based JPEG: computation is greatly reduced, image edges are better encoded, blocking is eliminated, and it allows lossless coding.

A subspace reverse-correlation technique for the study of visual neurons.

A new discrete-time reverse-correlation scheme for the study of visual neurons is proposed. The visual stimulus is generated by drawing with uniform probability, at each refresh time, an image from a finite set S of orthonormal images. We show that if the neuron can be modeled as a spatiotemporal linear filter followed by a static nonlinearity, the cross-correlation between the input image sequence and the cell's spike train output gives the projection of the receptive field onto the subspace spanned by S. The technique has been applied to the analysis of simple cells in the primary visual cortex of cats and macaque monkeys. Experimental results are presented where S spans a subspace of spatially low-pass signals. Advantages of the proposed scheme over standard white-noise techniques include improved signal to noise ratios, increased spatial resolution, and the possibility to restrict the study to particular subspaces of interest.

Microphone-array hearing aids with binaural output .I. Fixed-processing systems

This work is aimed at developing a design for the use of a microphone array with binaural hearing aids. The goal of such a hearing aid is to provide both the spatial-filtering benefits of the array and the natural benefits to sound localization and speech intelligibility that accrue from binaural listening. The present study examines two types of designs for fixed-processing systems: one in which independent arrays provide outputs to the two ears, and another in which the binaural outputs are derived from a single array. For the latter, various methods are used to merge array processing with binaural listening. In one approach, filters are designed to satisfy a frequency-dependent trade between directionality and binaural cue fidelity. In another, the microphone signals are filtered into low- and high-frequency components with the lowpass signals providing binaural cues and the highpass signal being the single output of the array processor. Acoustic and behavioral measurements were made in an anechoic chamber and in a moderately reverberant room to evaluate example systems. Theoretical performance was calculated for model arrays mounted on an idealized spherical head. Results show that both single- and dual-array systems provided target-intelligibility enhancements (2-4 dB improvements in speech reception threshold) relative to binaural cardioid microphones. In addition, the binaural-output systems provided cues that assist in sound localization, with resulting performance depending directly upon the cue fidelity. Finally, the sphere-based calculations accurately reflected the major features of the actual head-mounted array results, both in terms of directional sensitivity and output binaural cues.

A model of colour processing in the retina of vertebrates: From photoreceptors to colour opposition and colour constancy phenomena

Starting with a biological model of the foveal and parafoveal regions of the retina, this paper shows that simple considerations about the spatiotemporal filtering processed by the retinal neural network can account for colour processing at the level of early vision. We establish that the Red, Green, Blue signal can be considered as a low-pass luminance signal plus a colour-modulated signal. The structure of the Outer- and Inner Plexiform Layers of the retina leads to spatial low- and high-pass filters which account for the achromatic and transient characteristics of the Y ganglion cells as well as for the spatiotemporal colour-opponent properties of X ganglion cells. Considering this property and the logarithmic transduction of photoreceptors, it is easy to postulate that, after the retina at the level of the Lateral Geniculate Nucleus, a simple low-pass filtering can pave the way to the well known colour constancy phenomenon.

On-line interpolation using spline functions

An off-line or block B-spline interpolation algorithm is modified to an on-line or running form which handles growing data sequences more efficiently. Some loss of interpolation accuracy is suffered with lowpass signals but when the signal is broadband, the increase in interpolation error is negligible while a significant saving in computational effort is obtained.

Novel universal voltage-mode biquad filter with three inputs and one output using only two current conveyors

A new universal voltage-mode second-order filter with three inputs and one output employing two current conveyors, two capacitors and three resistors is presented. With the same number of passive elements, the new filter employs one less current conveyor than the latest published paper. Also, the new filter does not need a voltage follower. The new circuit still offers the following advantageous features of the published filter of this category: realization of all-pass, notch, high-pass, bandpass and low-pass signals from the same configuration, no requirements for component-matching conditions, orthogonal control of ω0 and Q, low active and passive sensitivities.