Showing papers on "Impulse response published in 2005"

A Practitioner's Guide to Lag Order Selection For VAR Impulse Response Analysis

Abstract: It is common in empirical macroeconomics to fit vector autoregressive (VAR) models to construct estimates of impulse responses. An important preliminary step in impulse response analysis is the selection of the VAR lag order. In this paper, we compare the six lag-order selection criteria most commonly used in applied work. Our metric is the mean-squared error (MSE) of the implied pointwise impulse response estimates normalized relative to their MSE based on knowing the true lag order. Based on our simulation design we conclude that for monthly VAR models, the Akaike Information Criterion (AIC) tends to produce the most accurate structural and semi-structural impulse response estimates for realistic sample sizes. For quarterly VAR models, the Hannan-Quinn Criterion (HQC) appears to be the most accurate criterion with the exception of sample sizes smaller than 120, for which the Schwarz Information Criterion (SIC) is more accurate. For persistence profiles based on quarterly vector error correction models with known cointegrating vector, our results suggest that the SIC is the most accurate criterion for all realistic sample sizes.

Use of modulated excitation signals in medical ultrasound. Part II: design and performance for medical imaging applications

Abstract: For pt.I, see ibid., vol.52, no.2, p.177-91 (2005). In the first paper, the superiority of linear FM signals was shown in terms of signal-to-noise ratio and robustness to tissue attenuation. This second paper in the series of three papers on the application of coded excitation signals in medical ultrasound presents design methods of linear FM signals and mismatched filters, in order to meet the higher demands on resolution in ultrasound imaging. It is shown that for the small time-bandwidth (TB) products available in ultrasound, the rectangular spectrum approximation is not valid, which reduces the effectiveness of weighting. Additionally, the distant range sidelobes are associated with the ripples of the spectrum amplitude and, thus, cannot be removed by weighting. Ripple reduction is achieved through amplitude or phase predistortion of the transmitted signals. Mismatched filters are designed to efficiently use the available bandwidth and at the same time to be insensitive to the transducer's impulse response. With these techniques, temporal sidelobes are kept below 60 to 100 dB, image contrast is improved by reducing the energy within the sidelobe region, and axial resolution is preserved. The method is evaluated first for resolution performance and axial sidelobes through simulations with the program Field II. A coded excitation ultrasound imaging system based on a commercial scanner and a 4 MHz probe driven by coded sequences is presented and used for the clinical evaluation of the coded excitation/compression scheme. The clinical images show a significant improvement in penetration depth and contrast, while they preserve both axial and lateral resolution. At the maximum acquisition depth of 15 cm, there is an improvement of more than 10 dB in the signal-to-noise ratio of the images. The paper also presents acquired images, using complementary Golay codes, that show the deleterious effects of attenuation on binary codes when processed with a matched filter, also confirmed by the presented simulated images.

Correlation-based decision-feedback equalizer for underwater acoustic communications

Abstract: The purpose of this paper is to develop a decision-feedback equalizer (DFE) using a fixed set of parameters applicable to most shallow oceans with minimal user supervision (i.e., a turn key system). This work is motivated by the superior performance [bit error rate (BER)] of the multichannel DFE compared with other methods, such as passive-phase conjugation (PPC), at the same time noting its sensitivity to different acoustic environments. The approach is to couple PPC, utilizing its adaptability to different environments, with a single-channel DFE. This coupling forms an optimal processor for acoustic communications in theory, but it has never been implemented in practice. By coupling with DFE, the method achieves the same spatial diversity as conventional multichannel DFE, without requiring a large number of receivers as does PPC. The correlation-based DFE in terms of the autocorrelation functions of the channel impulse responses summed over the receiver channels (the Q function) is derived. This paper shows in terms of waveguide physics, further supported by real data, the many desirable features of the Q function that suggest, given adequate sampling of the water column, a general applicability of the correlation-based equalizer to different environments, irrespective of the sound speed profiles, bottom properties, and source-receiver ranges/depths. This property can be expected to hold approximately for a small number of receivers with spatial diversity. This paper demonstrates the robustness of the new equalizer with moving source data despite the range change (which modifies the impulse response) and symbol phase change due to time-varying Doppler

Discrete Signals and Inverse Problems: An Introduction for Engineers and Scientists

Abstract: Preface. Brief Comments on Notation. 1. Introduction. 1.1 Signals, Systems, and Problems. 1.2 Signals and Signal Processing -- Application Examples. 1.3 Inverse Problems -- Application Examples. 1.4 History -- Discrete Mathematical Representation. 1.5 Summary. Solved Problems. Additional Problems. 2. Mathematical Concepts. 2.1 Complex Numbers and Exponential Functions. 2.2 Matrix Algebra. 2.3 Derivatives -- Constrained Optimization 2.4 Summary. Further Reading. Solved Problems. Additional Problems. 3. Signals and Systems. 3.1 Signals: Types and Characteristics. 3.2 Implications of Digitization -- Aliasing 3.3 Elemental Signals and Other Important Signals. 3.4 Signal Analysis with Elemental Signals. 3.5 Systems: Characteristics and Properties. 3.6 Combination of Systems 3.7 Summary. Further Reading. Solved Problems. Additional Problems. 4. Time Domain Analyses of Signals and Systems. 4.1 Signals and Noise. 4.2 Cross- and Autocorrelation: Identifying Similarities. 4.3 The Impulse Response -- System Identification. 4.4 Convolution: Computing the Output Signal. 4.5 Time Domain Operations in Matrix Form. 4.6 Summary. Further Reading. Solved Problems. Additional Problems. 5. Frequency Domain Analysis of Signals (Discrete Fourier Transform). 5.1 Orthogonal Functions -- Fourier Series. 5.2 Discrete Fourier Analysis and Synthesis. 5.3 Characteristics of the Discrete Fourier Transform. 5.4 Computation in Matrix Form. 5.5 Truncation, Leakage, and Windows. 5.6 Padding. 5.7 Plots. 5.8 The Two-dimensional Discrete Fourier Transform. 5.9 Procedure for Signal Recording. 5.10 Summary. Further Reading and References. Solved Problems. Additional Problems. 6. Frequency Domain Analysis of Systems. 6.1 Sinusoids and Systems - Eigenfunctions. 6.2 Frequency Response. 6.3 Convolution. 6.4 Cross-spectral and Autospectral Densities. 6.5 Filters in the Frequency Domain -- Noise Control. 6.6 Determining H with Noiseless Signals (Phase Unwrapping). 6.7 Determining H with Noisy Signals (Coherence). 6.8 Summary. Further Reading and References. Solved Problems. Additional Problems. 7. Time Variation and Nonlinearity. 7.1 Nonstationary Signals: Implications. 7.2 Nonstationary Signals: Instantaneous Parameters. 7.3 Nonstationary Signals: Time Windows. 7.4 Nonstationary Signals: Frequency Windows. 7.5 Nonstationary Signals: Wavelet Analysis. 7.6 Nonlinear Systems: Detecting Nonlinearity. 7.7 Nonlinear Systems: Response to Different Excitations. 7.8 Time-varying Systems. 7.9 Summary. Further Reading and References Solved Problems. Additional Problems. 8. Concepts in Discrete Inverse Problems. 8.1 Inverse Problems -- Discrete Formulation. 8.3 Data-driven Solution -- Error Norms. 8.4 Model Selection -- Ockham's Razor. 8.5 Information. 8.6 Data and Model Errors. 8.7 Nonconvex Error Surfaces. 8.8 Discussion on Inverse Problems. 8.9 Summary. Further Reading and References. Solved Problems. Additional Problems. 9. Solution by Matrix Inversion. 9.1 Pseudoinverse. 9.2 Classification of Inverse Problems. 9.3 Least Squares Solution (LSS). 9.4 Regularized Least Squares Solution (RLSS). 9.5 Incorporating Additional Information. 9.6 Solution Based on Singular Value Decomposition. 9.7 Nonlinearity. 9.8 Statistical Concepts -- Error Propagation. 9.9 Experimental Design for Inverse Problems. 9.10 Methodology for the Solution of Inverse Problems. 9.11 Summary. Further Reading. Solved Problems. Additional Problems. 10. Other Inversion Methods. 10.1 Transformed Problem Representation. 10.2 Iterative Solution of System of Equations. 10.3 Solution by Successive Forward Simulations. 10.4 Techniques from the Field of Artificial Intelligence. 10.5 Summary. Further Reading. Solved Problems. Additional Problems. 11. Strategy for Inverse Problem Solving. 11.1 Step 1: Analyze the Problem. 11.2 Step 2: Pay Close Attention to Experimental Design. 11.3 Step 3: Gather High-quality Data. 11.4 Step 4: Pre-process the Data. 11.5 Step 5: Select an Adequate Physical Model. 11.6 Step 6: Explore Different Inversion Methods. 11.7 Step 7: Analyze the Final Solution. 11.8 Summary. Solved Problems. Additional Problems. Index.

Unification and evaluation of equalization structures and design algorithms for discrete multitone modulation systems

Abstract: To ease equalization in a multicarrier system, a cyclic prefix (CP) is typically inserted between successive symbols. When the channel order exceeds the CP length, equalization can be accomplished via a time-domain equalizer (TEQ), which is a finite impulse response (FIR) filter. The TEQ is placed in cascade with the channel to produce an effective shortened impulse response. Alternatively, a bank of equalizers can remove the interference tone-by-tone. This paper presents a unified treatment of equalizer designs for multicarrier receivers, with an emphasis on discrete multitone systems. It is shown that almost all equalizer designs share a common mathematical framework based on the maximization of a product of generalized Rayleigh quotients. This framework is used to give an overview of existing designs (including an extensive literature survey), to apply a unified notation, and to present various common strategies to obtain a solution. Moreover, the unification emphasizes the differences between the methods, enabling a comparison of their advantages and disadvantages. In addition, 16 different equalizer structures and design procedures are compared in terms of computational complexity and achievable bit rate using synthetic and measured data.

Model order reduction using response-matching technique

Abstract: A method for model order reduction is proposed using response-matching technique. The step and impulse inputs have been considered. All types of pole configurations in the original high-order and reduced low-order system are included in this paper like real, complex and repeated. The proposed method is comparable in quality with similar existing methods and is capable of generating a reduced-order model with a desired pole pattern.

Performance analysis of the deficient length LMS adaptive algorithm

Abstract: In almost all analyzes of the least mean-square (LMS) finite impulse response (FIR) adaptive algorithm, it is assumed that the length of the adaptive filter is equal to that of the unknown system impulse response. However, in many practical situations, a deficient length adaptive filter, whose length is less than that of the unknown system, is employed, and analysis results for the sufficient length LMS algorithm are not necessarily applicable to the deficient length case. Therefore, there is an essential need to accurately quantify the behavior of the LMS algorithm for realistic situations where the length of the adaptive filter is deficient. In this paper, we present a performance analysis for the deficient length LMS adaptive algorithm for correlated Gaussian input data and using the common independence assumption. Exact expressions that completely characterize the transient and steady-state mean-square performances of the algorithm are developed, which lead to new insights into the statistical behavior of the deficient length LMS algorithm. Simulation experiments illustrate the accuracy of the theoretical results in predicting the convergence behavior of the algorithm.

Time-domain implementation of fractional order controllers

Abstract: Fractional order derivatives may be implemented in the time domain for control purposes in many ways. An exhaustive study of the performance of 28 different alternatives is presented and compared with discretised Crone controllers. Placement of zeros and poles, impulse and step responses and frequency responses are addressed. No formula is clearly better than all others in all situations, but it is expected that this survey will allow the choosing of the best possibility for each particular case.

Suppressing acoustic echo in a spectral envelope space

Abstract: Full-duplex hands-free telecommunication systems employ an acoustic echo canceler (AEC) to remove the undesired echoes that result from the coupling between a loudspeaker and a microphone. Traditionally, the removal is achieved by modeling the echo path impulse response with an adaptive finite impulse response (FIR) filter and subtracting an echo estimate from the microphone signal. It is not uncommon that an adaptive filter with a length of 50-300 ms needs to be considered, which makes an AEC highly computationally expensive. In this paper, we propose an echo suppression algorithm to eliminate the echo effect. Instead of identifying the echo path impulse response, the proposed method estimates the spectral envelope of the echo signal. The suppression is done by spectral modification-a technique originally proposed for noise reduction. It is shown that this new approach has several advantages over the traditional AEC. Properties of human auditory perception are considered, by estimating spectral envelopes according to the frequency selectivity of the auditory system, resulting in improved perceptual quality. A conventional AEC is often combined with a post-processor to reduce the residual echoes due to minor echo path changes. It is shown that the proposed algorithm is insensitive to such changes. Therefore, no post-processor is necessary. Furthermore, the new scheme is computationally much more efficient than a conventional AEC.

A, b,c's (and d)'s for understanding vars

Abstract: The dynamics of a linear (or linearized) dynamic stochastic economic model can be expressed in terms of matrices (A, B,C, D) that define a state space system. An associated state space system (A, K,C, Σ) determines a vector autoregression for observ- ables available to an econometrician. We review circumstances under which the impulse response of the VAR resembles the impulse response associated with the economic model. We give four examples that illustrate a simple condition for checking whether the mapping from VAR shocks to economic shocks is invertible. The condition applies when there are equal numbers of VAR and economic shocks.

Extensive Development of Leak Detection Algorithm by Impulse Response Method

Abstract: The oscillatory flows in pipeline systems due to excitation by valve operation are efficiently analyzed by the impulse response method. The impact of leakage is incorporated into the transfer functions of the complex head and discharge. Frequency-dependent friction is used to consider the impact of unsteady friction for laminar condition. Extensive development of the impulse response method was made by considering the sources of friction associated with the local and convective acceleration of velocity for turbulent flow. The genetic algorithm was integrated into the impulse response method to calibrate the location and the quantity of leakage. The calibration function for leakage detection can be made using the pressure-head response at the valve, or the pressure-head and flow response at the section upstream from the valve. The proposed leak detection algorithm shows the potentials for being applied to a simple pipeline system with a single leak or multiple leaks.

Staggered pilot transmission for channel estimation and time tracking

Abstract: To allow a receiving entity to derive a longer channel estimate while limiting overhead, a transmitting entity transmits a pilot on different groups of subbands in different time intervals. N subbands in the system are arranged into M non-overlapping groups. Each group includes P=N/M subbands that are uniformly distributed across the N subbands. The transmitting entity transmits the pilot on a different subband group in each time interval, and selects all M subband groups in M time intervals based on a pilot staggering pattern. The receiving entity derives (1) an initial impulse response estimate with P channel taps based on the pilot received on one subband group and (2) two longer impulse response estimates with different lengths used for data detection and time tracking. Each longer impulse response estimate may be derived by filtering initial impulse response estimates for a sufficient number of subband groups using a time-domain filter.

Joint detection-estimation of brain activity in functional MRI: a Multichannel Deconvolution solution

Abstract: Analysis of functional magnetic resonance imaging (fMRI) data focuses essentially on two questions: first, a detection problem that studies which parts of the brain are activated by a given stimulus and, second, an estimation problem that investigates the temporal dynamic of the brain response during activations. Up to now, these questions have been addressed independently. However, the activated areas need to be known prior to the analysis of the temporal dynamic of the response. Similarly, a typical shape of the response has to be assumed a priori for detection purpose. This situation motivates the need for new methods in neuroimaging data analysis that are able to go beyond this unsatisfactory tradeoff. The present paper raises a novel detection-estimation approach to perform these two tasks simultaneously in region-based analysis. In the Bayesian framework, the detection of brain activity is achieved using a mixture of two Gaussian distributions as a prior model on the "neural" response levels, whereas the hemodynamic impulse response is constrained to be smooth enough in the time domain with a Gaussian prior. All parameters of interest, as well as hyperparameters, are estimated from the posterior distribution using Gibbs sampling and posterior mean estimates. Results obtained both on simulated and real fMRI data demonstrate first that our approach can segregate activated and nonactivated voxels in a given region of interest (ROI) and, second, that it can provide spatial activation maps without any assumption on the exact shape of the Hemodynamic Response Function (HRF), in contrast to standard model-based analysis.

Human Identification System Based ECG Signal

Abstract: A new human identification system based electrocardiogram (ECG) signal is introduced in this work. The human heart is considered to be a unique system of each person. ECG signal therefore represents as an impulse response of the system. The frequency response of the system (Fourier transform of the ECG signal) is employed to be a tool for feature extraction. In addition, the ECG signals employed in this paper, which may be derived from different heart rates from different subjects at the recording time, are normalized to a standard heart rate. Furthermore, not only the whole sequence of 1 period EC signal, containing P, QRS, and T waves, is processed but also its three subsequences, each respectively representing P, QRS, and T waves, is examined. The results of using neural network have demonstrated that subsequences technique is superior to whole period of ECG signal method.

In situ characterization of building materials for propagation modeling: frequency and time responses

Abstract: We propose original data processing methods for the dielectric characterization of frequency-dependent reflection coefficients of construction materials considering a very wide frequency band. Two types of approaches have been developed to obtain, from spectral measurements, estimates of the equivalent complex permittivity versus frequency or reconstruction of the impulse response. In particular, high-resolution (HR) algorithms based on the matrix pencil method have been used in an original way to identify wave multipath inside a sample. Both approaches have been used for the characterization of different types of building materials. A database of dielectric responses of materials is under construction in order to provide the deterministic propagation simulator with the characteristics of building materials.

Error analysis of the simulated impulse response on indoor wireless optical channels using a Monte Carlo-based ray-tracing algorithm

Abstract: This work describes a method to determine the error in a Monte Carlo-based ray-tracing algorithm used to compute the impulse response on indoor wireless optical channels. The algorithm, which accounts for multiple reflections of any order on irregularly shaped furnished rooms with diffuse and specular reflectors, allows for their analysis. Equations that estimate algorithm-produced error are given. We also report several simulation results concerning the error estimation which verify the reliability of the equations.

Causality, Stokes' wave equation, and acoustic pulse propagation in a viscous fluid.

Abstract: Stokes' acoustic wave equation is solved for the impulse response of an isotropic viscous fluid. Two exact integral forms of solution are derived, both of which are causal, predicting a zero response before the source is activated at time t = 0. Moreover, both integral solutions satisfy a stronger causality condition: the pressure pulse is maximally flat, with all its time derivatives identically zero at t = 0, signifying that there is no instantaneous response to the source anywhere in the fluid. A closed-form approximation for each of the two integrals is derived, with distinctly different properties in the two cases, even though the original integrals are equivalent in that they predict identical pulse shapes. One of these approximations, reminiscent of transient solutions that have appeared previously in the literature, is noncausal due to the incorrect representation of high-frequency components in the propagating pulse. In the second approximation, all frequency components are treated correctly, leading to an impulse response that satisfies the strong causality condition, also satisfied by the original integrals, whereby the predicted pressure pulse is zero when t < 0 and maximally flat everywhere in the fluid immediately after t = 0.

Receiver circuit comprising equalizer

Abstract: A receiver circuit has an equalizer that equalizes a received signal propagating through a transmission medium; a data detection circuit that detects an analog output signal of the equalizer at a data sample timing and outputs a digital signal; an intersymbol interference detection circuit that detects an intersymbol interference level from the analog output signal of the equalizer at the data sample timing and from the digital signal of the data detection circuit; and an equalization characteristic control unit that controls the characteristic of the equalizer to minimize the detected intersymbol interference level The receiver circuit further has a data sample timing control unit in which the data sample timing is controlled to a sample timing at which the difference between the amplitude of the analog output waveform of the equalizer with respect to an impulse and the amplitude of an ideal impulse response waveform is minimal

Efficient reduced complexity windowed optimal time domain equalizer for discrete multitone-based DSL modems

Abstract: An algorithm for computing an efficient, reduced complexity, windowed optimal linear time domain equalizer for a dispersive channel comprises the steps of determining a window of maximum energy in the impulse response of length equal to or less than a number of cyclic prefix samples associated with a received digital data signal, computing the corresponding inside and outside matrices, performing an inverse Cholesky decomposition of the inside matrix, creating a resultant matrix as the product of the outer and the upper and lower square root inner matrix, followed by Householder reduction and QL transformation to thereby compute the time domain equalizer as the linear transformation of the eigenvector corresponding to the smallest eigenvalue at the receiver. The smallest eigenvalue is determined using the aforementioned orthogonal transformations without determining all the eigenvalues efficiently but without the loss accuracy associated with iterative methods like the conventional power method. The algorithm may be most conveniently implemented, for example, in the form of a thirty-two bit digital signal processor at a data receiver.

Demonstrating Time Reversal in Ultra-wideband Communications Using Time Domain Measurements

Abstract: In Time- Reversal (TiR), a signal is pre- filtered such that it focuses in space and time at an intended receiver. This can be achieved by using a time-reversed complex conjugate of the channel impulse response at the receiver as a transmitter pre- filter. Several advantages come with this technique. Spatial focusing reduces co-channel interference in a multi-user system. Due to temporal focusing, the effective delay spread of the channel is dramatically reduced and the complexity of the receiver is thus reduced. In this paper, we describe a series of measurement campaign carried out to illustrate the space-time focusing properties of TiR. Ultra- wideband (UWB) channels are measured by sounding the channels with a sub-nanosecond pulse. CLEAN algorithm is used to extract the channel impulse response. From the observed impulse response, the leverages of TR in UWB are then demonstrated.

Discrete Green's function formulation of the FDTD method and its application in antenna modeling

Abstract: A discrete Green's function formulation of the finite-difference time-domain (DGF-FDTD) method based on both discrete system theory and the FDTD method has been developed, which expresses the field response as a convolution of current sources and the impulse response of the FDTD equation system. The DGF-FDTD method presents the FDTD equations in a different perspective from the conventional Yee algorithm. It avoids the computational difficulties such as the need for computation of free-space nodes and absorbing boundary conditions of the classic FDTD method. The ability of the DGF-FDTD method to model on antenna is demonstrated by the modeling of a Yagi-Uda array antenna with considerable saving in memory usage.

51.1: Temporal Impulse Response and Bandwidth of Displays in Relation to Motion Blur

Abstract: Displays with non-impulse temporal response suffer from motion blur artifacts. This paper will show that the temporal impulse response and related temporal bandwidth provide a simple and direct way to quantify motion blur.

A Wavelet Packet Based Sifting Process and its Application for Structural Health Monitoring

Abstract: This article presents an innovative wavelet packet based sifting process to decompose a signal into its components with different frequency contents by examining the energy content in the wavelet packet components of a signal and imposing certain decomposition criteria. A new method is illustrated for simulation data of a linear three-degree-of-freedom spring-mass-damper system and the results are compared with those obtained using the empirical mode decomposition (EMD) method. Both methods provide good approximations, as compared with the exact solution for modal responses from a conventional modal analysis and both show relatively greater errors at the beginning and ending parts of the signal due to the well-known end effects. A comparison study is also provided to illustrate differences between two sifting processes of the proposed approach and the EMD, by using a harmonic signal with a sweeping frequency and the impulse response of a linear single-degree-of-freedom system with viscous damping. Incorpo...

Optimized temperature modulation of micro-hotplate gas sensors through pseudorandom binary sequences

Abstract: In recent years, modulating the working temperature of metal-oxide gas sensors has been one of the most widely used methods to enhance sensor selectivity. When the working temperature of a gas sensor is modulated, the kinetics of the gas-sensor interaction are altered, and this leads to characteristic response patterns. Many works have shown that it is possible to identify and determine the concentration of gases in simple mixtures, even using a single temperature-modulated metal-oxide gas sensor. However, the selection of the frequencies used to modulate temperature remains an empirical process. In this paper, we introduce a method, borrowed from the field-of-system identification, to systematically determine the optimal set of modulating frequencies to solve a given gas-analysis application. The method consists of using maximum-length pseudorandom binary sequences to modulate the working temperature of metal-oxide gas sensors. Since these signals have a flat power spectrum (i.e., like white noise) in a wide frequency range, an estimate of the impulse response of each gas-sensor pair can be computed by the cross correlation of the excitatory and response sequences. Studying the impulse response estimates, the set of modulating frequencies that are useful to discriminate between different gases and to estimate gas concentration, is obtained in a systematic way. The method is demonstrated with tungsten oxide micro-hotplate gas sensors applied to detect ammonia, nitrogen dioxide, and their binary mixtures at different concentrations. It is shown that it is possible to find temperature-modulating frequencies to obtain high gas identification and quantification rates (95.55% and 100%, respectively).

Impulse responses of linear UWB antenna arrays and the application to beam steering

Abstract: Ultra wideband (UWB) systems enable the transmission of short pulses with a very high time resolution The application of UWB arrays leads to small beam widths without sidelobes even for sparse arrays with large element spacing However the transmitted and received signals are subject to distortions due to the dispersive properties like ringing and filtering of the antennas These effects are analyzed for true time delay beam steering in the paper by the means of the antenna array impulse response This is derived for the example of a linear array for the frequency range of 31-106 GHz The theoretical results are verified by measurements

Estimation of frequency-averaged loss factors by the power injection and the impulse response decay methods.

Abstract: This paper describes a comparison, both analytically and experimentally, between two widely used loss factor estimation techniques frequently used in statistical energy analysis Analytical models of simple spring/mass/damper systems were created to compare frequency-averaged loss factor values from the single subsystem power injection method and the impulse response decay method The parameters of the analytical models were varied to study the effects of the total number of modes, amount of damping, location of modes within frequency bands, and the width of the frequency bands on loss factor estimation The analytical study shows that both methods give accurate loss factor values as long as the damping values remain realistic for linear systems and at least one modal resonance is present in each frequency band These analytical results were verified experimentally by measuring the loss factors of simple steel plates, with and without damping treatments applied

Blind and semi-blind equalization based on the constant power criterion

Abstract: This paper focuses on the constant power (CP) criterion for blind linear equalization of digital communication channels. This recently proposed criterion is specially designed for the extraction of q-ary phase shift keying (q-PSK) signals using finite impulse response equalizers. When zero-forcing equalizers exist, the CP cost function accepts exact analytic solutions that are unaffected by undesired local extrema and spare costly iterative optimization. A subspace-based method exploiting the Toeplitz-like structure of the solution space is put forward to recover the minimum-length equalizer impulse response from the overestimated-length solutions. The proposed method is more robust to the relative weights of the minimum-length equalizer taps than existing techniques. In less ideal scenarios where the analytic solutions are only approximate minimizers of the criterion, a gradient-descent algorithm is proposed to minimize the cost function. To reduce the detrimental effects of suboptimal equilibria and accelerate convergence, the iterative algorithm is initialized with the approximate closed-form solution, and an optimal step size is incorporated into its updating rule. This optimal step size, which globally minimizes the cost function along the search direction, can be computed algebraically. A semi-blind implementation, which is useful when training data are available, further reduces the impact of undesired local extrema and enhances the convergence characteristics (particularly the robustness to the equalizer initialization) of the iterative algorithm from just a few pilot symbols. All these beneficial features are demonstrated with an experimental study of the proposed CP-based methods in a variety of channels and simulation conditions.