Anti-aliasing Filter Research Articles

MOCK CIRCULATORY SYSTEM FOR THE CONTINUOUS FLOW TOTAL ARTIFICIAL HEART

Purpose: A mock circulatory loop was designed to be used as a platform for testing feedback control strategies for the continuous flow total artificial heart (TAH). Methods: The mock circuit comprises both systemic and pulmonary circulations. The TAH consists of 2 axial flow pumps. The atria are flexible and collapse under excessive negative pressure. Pulmonary and systemic vascular resistances are adjusted to approximately 4 and 20 Wood units, respectively, using tubing clamps. The respective compliances are modeled with sealed cylinders containing air. During normal operation, approximately 80% of the fluid volume is in the systemic circulation. Pressure signals are filtered with anti-aliasing filters. Aortic and pulmonary arterial pressures, left and right atrial pressures, and flow rates are sampled at 1 kHz. Results: A wide range of hemodynamic conditions were simulated. Baseline aortic and pulmonary arterial pressures were maintained at 80 and 20 mm Hg, respectively, while atrial pressures were kept at approximately 0 mm Hg. Flow rate in the left pump was slightly higher than in the right due to the bronchial shunt. The mock circuit was successfully used for testing physiologic feedback control strategies for the dual-pump system. Conclusions: Normal and pathologic conditions, including atrial collapse, can be readily simulated in the mock circuit, making it a suitable platform for testing feedback control algorithms aimed at correcting hemodynamic abnormalities in patients with TAHs.

Improved line spectral frequency estimation through anti-aliasing filtering

A study of the classic line spectral frequency (LSF) extraction methods along with the assumptions made during their estimation is presented. LSF extraction is investigated from an over-sampling and decimation perspective and LSFs are shown to contain high-frequency variations that led to spectral overlapping problems. An anti-aliasing filter prior to decimation, with a cut-off frequency dependent on the final LSF vector transmission rate, is proposed to alleviate the aliasing problem of the classic extraction methods. The proposed method shows a clear advantage because it produces the same quantisation distortion as the classic methods at a lower bit requirement, with significant reduction in 2 and 4 dB outliers that greatly affect synthesised speech quality. This was also confirmed through a listening test.

Application of undersampling technique for the design of an NMR signals digital receiver

In this article, undersampling technique theory and design, realization, and workbench test of a low-cost digital MRI (magnetic resonance imaging) receiver are re- ported. The authors first discuss classic analog receiver architecture, reporting its disad- vantages, and then present a receiver with a complete description of the electronic circuit. The intended use of the subsystem should be as a radiofrequency (RF) receiver chain in a low-cost dedicated MRI scanner (e.g., suitable for musculoskeletal limbs studies). The digital receiver consists of a passband (antialiasing) filter, an ADC (analog to digital converter) to digitalize the signal with the undersampling technique, and a DDC (digital down converter) for the frequency translation and filtering to process the signal using a PC. The use of the DDC guarantees perfect signals quadrature and high-performance filtering; moreover, it is adaptable to each modification of the system because of its programmabil- ity. Detailed specifications and hardware design of the digital receiver that is designed to be used in a dedicated MR scanner is provided, guaranteeing top performance and low cost. © 2006 Wiley Periodicals, Inc. Concepts Magn Reson Part B (Magn Reson Engineering) 29B: 107-114, 2006

Analysis of single ion channel data incorporating time-interval omission and sampling

Hidden Markov models are widely used to describe single channel currents from patch-clamp experiments. The inevitable anti-aliasing filter limits the time resolution of the measurements and therefore the standard hidden Markov model is not adequate anymore. The notion of time-interval omission has been introduced where brief events are not detected. The developed, exact solutions to this problem do not take into account that the measured intervals are limited by the sampling time. In this case the dead-time that specifies the minimal detectable interval length is not defined unambiguously. We show that a wrong choice of the dead-time leads to considerably biased estimates and present the appropriate equations to describe sampled data.

Flexible Analog Front Ends of Reconfigurable Radios Based on Sampling and Reconstruction with Internal Filtering

Bandpass sampling, reconstruction, and antialiasing filtering in analog front ends potentially provide the best performance of software defined radios. However, conventional techniques used for these procedures limit reconfigurability and adaptivity of the radios, complicate integrated circuit implementation, and preclude achieving potential performance. Novel sampling and reconstruction techniques with internal filtering eliminate these drawbacks and provide many additional advantages. Several ways to overcome the challenges of practical realization and implementation of these techniques are proposed and analyzed. The impact of sampling and reconstruction with internal filtering on the analog front end architectures and capabilities of software defined radios is discussed.

발사체 탑재용 파이로 충격기록장치 개발 및 시험

Pyro-shock measurement system in launch vehicle is necessary to validate structural robustness and to prepare environmental qualification criteria for launch vehicle systems. This paper describes design, development and function test of prototype pyro-shock recorder to be employed on KSLV-I in the near future. Due to the limitation of telemetry transmission rate, pyro-shock recorder acquires and stores the shock sensor data with high sampling rate in short period and sends the data to the KSLV-I telemetry system with lower data rate. Signal conditioning in pyro-shock recorder is designed to enhance signal-to-noise ratio through proper placement of anti-aliasing filter.

A Subsequent Fit of Time Series and Amplitude Histogram of Patch-Clamp Records Reveals Rate Constants up to 1 per Microsecond

Fast gating in time series of patch-clamp current demands powerful tools to reveal the rate constants of the adequate Hidden Markov model. Here, two approaches are presented to improve the temporal resolution of the direct fit of the time series. First, the prediction algorithm is extended to include intermediate currents between the nominal levels as caused by the anti-aliasing filter. This approach can reveal rate constants that are about 4 times higher than the corner frequency of the anti-aliasing filter. However, this approach is restricted to time series with very low noise. Second, the direct fit of the time series is combined with a beta fit, i.e., a fit of the deviations of the amplitude histogram from the Gaussian distribution. Since the "theoretical" amplitude histograms for higher-order Bessel filters cannot be calculated by analytical tools, they are generated from simulated time series. In a first approach, a simultaneous fit of the time series and of the Beta fit is tested. This simultaneous fit, however, inherits the drawbacks of both approaches, not the benefits. More successful is a subsequent fit: The fit of the time series yields a set of rate constants. The subsequent Beta fit uses the slow rate constants of the fit of the time series as fixed parameters and the optimization algorithm is restricted to the fast ones. The efficiency of this approach is illustrated by means of time series obtained from simulation and from the dominant K+ channel in Chara. This shows that temporal resolution can reach the microsecond range.

Time domain upsampling linear MMSE receiver for asynchronous MC-CDMA systems over frequency-selective Rayleigh fading channels

In this paper, a novel time domain upsampling linear receiver is proposed for an asynchronous multicarrier code division multiple access system in frequency-selective Rayleigh fading channels. After anti-aliasing filtering, the received discrete-time upsampled signal is processed by a finite-impulse-response (FIR) filter at the receiver without any pre-processing. For the FIR filter, the observation window is extended to two-symbol duration in order to capture a complete desired symbol and eliminate the knowledge of signature waveform and timing acquisition requirement, and the tap coefficients are optimised according to the minimum mean square error (MMSE) criterion by employing a normalised least-mean-square algorithm. The proposed time domain upsampling MMSE (TDU-MMSE) receiver works in a decision-directed mode after training by a pre-defined sequence. The theoretical bit error rate (BER) is derived for the proposed receiver and agrees very well with the simulation results. Moreover, it has been shown that the TDU-MMSE receiver is able to achieve better BER performance than the conventional linear MMSE (LMMSE) and partial sampling MMSE (PS-MMSE) with lower computation complexity.

Controlled behaviour of STF in CT modulators

One of the most interesting properties of continuous time /spl Sigma//spl Delta/ modulators is their implicit anti-aliasing filter. Unfortunately, peaking in their signal transfer function (STF) demands a 'no-overload' filter to precede the modulator. A technique and two new topologies, which can achieve an almost flat STF, are proposed. Meanwhile, the topologies still have the desired feature that the internal nodes contain few input related signals.

A11: Measurement system for data acquisition and electric arc furnace characteristic calculation

The paper presents the requirements of measurement system for electric arc furnace parameters measurement and for working characteristic calculating. On this basis measurement signals were specified and measurement algorithms were developed. For data acquisition a special system was constructed. The system consists of portable computer with acquisition card and conditioning system with differential amplifiers, antialiasing filters and sample and hold circuits. The system is connected with interface build-up permanent in arc furnace. The system software makes possible on-line observation of measured signals values and save data to hard disk. Further calculations may be performed off-line using special developed algorithms. For system software development the open source Python language and GCC compiler for Windows were used.

Novel Ideas for Fast Muscle Action Potential Simulations Using the Line Source Model

Using a signal processing approach, we analyze the line source model for muscle action potential (AP) modeling. We show that the original model presents a tradeoff between violating the Nyquist criterion on one hand and using a discretization frequency that is unnecessarily high with respect to the bandwidth of the generated AP on the other. Here, we present an improved line source model that, compared to the original, allows a lower discretization frequency while retaining the accuracy by simply introducing a continuous-time anti-aliasing filter. Moreover, a transfer function form of the transmembrane current is presented that promote the use of sophisticated signal processing methods on these type of signals. Both continuous-time and discrete-time models are presented. We also address and analyze the implications of the finite length of the muscle fibers. Including this in the model is straightforward, owing to the convolutional form of the line source model, and is manifested by a simple transformation of the associated weighting function. AP modeling is discussed for the three different electrode models: the concentric needle electrode, the single fiber electrode, and the macro electrode. The presented model is suitable for modeling large motor units, where both accuracy and computational efficiency are important factors. To simplify the selection of the discretization interval, we derive what we call the cumulative cutoff frequency that provides an estimate of the required Nyquist frequency.

Artifact-Suppressing Amplifier for EMG Registration during Repetitive Peripheral Magnetic Stimulation (RPMS)

To induce a most physiological proprioceptive inflow to the CNS, the intensity and the repetition frequency of the RPMS are adapted by a closed loop control to induce coordinated movements. For the closed loop control and the quantification of the therapeutic outcome it is necessary to measure the (remaining) voluntary muscle activity, while the RPMS is applied. This is essential, since the patient is advised to assist the induced movements as far as possible. Therefore an RPMS artefact-suppressing EMG amplifier has to be developed to measure the activity of the stimulated muscle during the stimulation. Hereby the stimulation coil is located directly above the two EMG electrodes. Such amplifiers are well-known and established for functional electrostimulation, while artefact-suppressing amplifiers for RPMS are fairly unknown. Because of RPMS impulses, a high (compared to the EMG) voltage is induced into the EMG electrodes. This voltage changes the charge of the skin capacity and therefore the DC offset of the EMG. The induced voltage also elicits an impulse response of the filters in the EMG amplifier. Hence an artefact suppression is necessary, otherwise the amplifier is permanently in saturation since the RPMS impulses are repeated with 20Hz. The trigger signal for the stimulation device is also used for the artefact suppression. The artefact suppression itself runs on three levels: Voltages higher than 1.4 V at the EMG electrodes are filtered by antiparallel diodes independent of the stimulation impulse. If the suppression is triggered, the input to the difference amplifier is shorted by two bipolar transistors and the input to the DC offset compensation is grounded. With this suppression unit its is possible to reduce the gap in the measured EMG down to 3 ms. In addition to the artefact suppression, the following requirements (excerpt) are fulfilled in the developed EMG amplifier: selectable amplification gain, bandwidth of 3Hz to 15 kHz, an anti-aliasing filter with a selectable edge frequency, a 50Hz band-stop filter and an active shielding (no external pre-amplifier module). Due to a difference amplifier and the artefact suppression the input resistance is at least 500 MOhm and the patient is protected form dangerous high voltages by an isolation barrier (3 kVrms). Also a conventional speaker module is integrated into the designed amplifier.

Multi-rate receivers with IF sampling and digital timing correction

This contribution deals with the design of a fully digital multi-rate receiver, operating on non-synchronized bandpass samples. Issues related to aliasing, sample rate conversion and timing correction are considered. We compare various receiver alternatives in terms of the BER degradation caused by aliasing and non-ideal timing correction. In particular, we focus on a low-complexity receiver performing sample rate conversion prior to matched filtering by means of an interpolator. We show that the BER degradation can be reduced to about 0.1 dB by careful selection of the system parameters, but at the expense of a large oversampling factor. A similar BER degradation can be achieved at a considerably smaller oversampling factor by including a simple digital anti-aliasing filter in front of the interpolator. We will develop a new type of anti-aliasing filter based on CIC filters. The distortion introduced by this CIC-type filter can be compensated for, by replacing the matched filter by an optimized receiver filter.

Time-domain Simulation of electronic noises

In this paper, a procedure is proposed to computer simulate the electronic noise of ionizing-radiation spectrometers. The viewpoint of the simulator is the output of the preamplifier, with or without an anti-aliasing filter, just in front of the ADC. Examples are given for the case of segmented high purity Germanium detectors (HPGe). The method makes use of the fractional calculus basics. A software procedure provides the noisy waveform as a function of the fundamental electrical-physical parameters of the system, including: detector capacitance, detector leakage current, feedback resistor, 1/f-noise coefficient of the input transistor, temperature of the preamplifier input devices. The ADC quantization noise is also included in the simulation.

A highly integrated dual-band multimode wireless LAN transceiver

A fully integrated radio transceiver chip for the 2.4- and 5-GHz WLAN standards 802.11a/b/g is presented in a 0.25-/spl mu/m 40-GHz BiCMOS technology. The chip integrates the low-noise amplifiers, mixers, channel filters, programmable gain control, synthesizers with voltage-controlled oscillators and reference oscillator, transmitters, antialiasing filters, and voltage regulators. The key performances of the presented transceiver are a receive sensitivity of -85 dBm and -74 dBm for 11-Mb/s complementary code keying (CCK) and 54 Mb/s orthogonal frequency division multiplexing (OFDM) modes, respectively, and an error vector magnitude of -35 dB measured at the transmitter with an output power of -4 dBm at 54-Mb/s 802.11a mode. The transceiver exceeds all IEEE requirements for the 802.11a/b/g CCK and OFDM standards and supports a frequency range of 4.9 to 6 GHz for the future extensions of the 802.11a standard in different countries.

Corrections to “Sampling and Signal Reconstruction Circuits Performing Internal Antialiasing Filtering and Their Influence on the Design of Digital Receivers and Transmitters”

Sampling and signal reconstruction circuits performing internal antialiasing filtering are derived and analyzed within the same paradigm. Their capabilities are evaluated. The practical realization of these circuits and their influence on the design of digital receivers and transmitters are considered.

MatLab program for precision calibration of optical tweezers

Optical tweezers are used as force transducers in many types of experiments. The force they exert in a given experiment is known only after a calibration. Computer codes that calibrate optical tweezers with high precision and reliability in the ( x, y)-plane orthogonal to the laser beam axis were written in MatLab (MathWorks Inc.) and are presented here. The calibration is based on the power spectrum of the Brownian motion of a dielectric bead trapped in the tweezers. Precision is achieved by accounting for a number of factors that affect this power spectrum. First, cross-talk between channels in 2D position measurements is tested for, and eliminated if detected. Then, the Lorentzian power spectrum that results from the Einstein–Ornstein–Uhlenbeck theory, is fitted to the low-frequency part of the experimental spectrum in order to obtain an initial guess for parameters to be fitted. Finally, a more complete theory is fitted, a theory that optionally accounts for the frequency dependence of the hydrodynamic drag force and hydrodynamic interaction with a nearby cover slip, for effects of finite sampling frequency (aliasing), for effects of anti-aliasing filters in the data acquisition electronics, and for unintended “virtual” filtering caused by the position detection system. Each of these effects can be left out or included as the user prefers, with user-defined parameters. Several tests are applied to the experimental data during calibration to ensure that the data comply with the theory used for their interpretation: Independence of x- and y-coordinates, Hooke's law, exponential distribution of power spectral values, uncorrelated Gaussian scatter of residual values. Results are given with statistical errors and covariance matrix. Program summary Title of program: tweezercalib Catalogue identifier: ADTV Program obtainable from: CPC Program Library, Queen's University of Belfast, N. Ireland. Program Summary URL: http://cpc.cs.qub.ac.uk/summaries/ADTV Computer for which the program is designed and others on which it has been tested: General computer running MatLab (MathWorks Inc.). Programming language used: MatLab (MathWorks Inc.). Uses “Optimization Toolbox” and “Statistics Toolbox”. Memory required to execute with typical data: Of order 4 times the size of the data file. High speed storage required: None No. of lines in distributed program, including test data, etc.: 133 183 No. of bytes in distributed program, including test data, etc.: 1 043 674 Distribution format: tar gzip file Nature of physical problem: Calibrate optical tweezers with precision by fitting theory to experimental power spectrum of position of bead doing Brownian motion in incompressible fluid, possibly near microscope cover slip, while trapped in optical tweezers. Thereby determine spring constant of optical trap and conversion factor for arbitrary-units-to-nanometers for detection system. Method of solution: Elimination of cross-talk between quadrant photo-diode's output channels for positions (optional). Check that distribution of recorded positions agrees with Boltzmann distribution of bead in harmonic trap. Data compression and noise reduction by blocking method applied to power spectrum. Full accounting for hydrodynamic effects: Frequency-dependent drag force and interaction with nearby cover slip (optional). Full accounting for electronic filters (optional), for “virtual filtering” caused by detection system (optional). Full accounting for aliasing caused by finite sampling rate (optional). Standard non-linear least-squares fitting. Statistical support for fit is given, with several plots suitable for inspection of consistency and quality of data and fit. Restrictions on the complexity of the problem: Data should be positions of bead doing Brownian motion while held by optical tweezers. For high precision in final results, data should be time series measured over a long time, with sufficiently high experimental sampling rate: The sampling rate should be well above the characteristic frequency of the trap, the so-called corner frequency. Thus, the sampling frequency should typically be larger than 10 kHz. The Fast Fourier Transform applied requires the time series to contain 2 n data points, and long measurement time is obtained with n>12–15. Finally, the optics should be set to ensure a harmonic trapping potential in the range of positions visited by the bead. The fitting procedure checks for harmonic potential. Typical running time: (Tens of) minutes Unusual features of the program: None

An Improved Multipath Channel Identification Algorithm Using Mulichannel Linear Prediction

Blind channel identification exploits the measurable channel output signal and some a priori knowledge of the statistics of the channel input signal. However, in many scenarios, more side information is available. In digital communication systems, the pulseshaping filter in the transmitter and the antialiasing filter in the receiver are often known to the receiver. Exploitation of this prior knowledge can simplify the channel identification problem. In this paper, we provide a thorough theoretical analysis on the linear prediction (LP)-based multipath identification approach. The analysis shows that the order overdetermination greatly deteriorates the performance of the LP multipath identification approach. This article, then, proposes an effective approach to detect the order of the multipath channel and alleviate the performance degradation attributed to the channel overmodeling. The obtained results demonstrate that the proposed approach offers substantial performance gain.

Focus on Data Acquisition

Focus on Data Acquisition

A CT Saturation Detection Algorithm

This paper proposes an algorithm for detecting current transformer (CT) saturation by analyzing a secondary current. The current has points of inflection where saturation begins and ends. Then, discontinuity in the first difference of the current arises at the next instants of points of inflection, where the second and third differences of the current may have large values enough to detect saturation. In this paper, the third difference is used because it is more effective than the second difference in terms of saturation detection. However, an anti-aliasing low-pass filter softens the current and, thus, reduces the values of the third difference at those instants. This paper includes the effect of a low-pass filter on the proposed algorithm. The sampling rate of 64 samples/cycle (s/c) is studied by varying a remanent flux and the cutoff frequency of the filter. The test results clearly indicate that the proposed algorithm successfully detects the interval of saturation even with a remanent flux and the softening effect. However, special attention should be paid to the decision of the threshold value when a cutoff frequency is much lower than half the sampling frequency. This paper concludes by implementing the algorithm into a TMS320C6701 digital signal processor and the results are satisfactory.