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Showing papers on "Impulse response published in 2009"


Proceedings ArticleDOI
05 Jul 2009
TL;DR: This paper describes a new database of binaural room impulse responses (BRIR), referred to as the Aachen Impulse Response (AIR), which covers a wide range of situations where digital hearing aids or other hands-free devices can be used.
Abstract: This paper describes a new database of binaural room impulse responses (BRIR), referred to as the Aachen Impulse Response (AIR) database. The main field of application of this database is the evaluation of speech enhancement algorithms dealing with room reverberation. The measurements with a dummy head took place in a low-reverberant studio booth, an office room, a meeting room and a lecture room. Due to the different dimensions and acoustic properties, it covers a wide range of situations where digital hearing aids or other hands-free devices can be used. Besides the description of the database, a motivation for using binaural instead of monaural measurements is given. Furthermore an example using a coherence-based dereverberation technique is provided to show the advantage of this database for algorithm evaluation. The AIR database is being made available online.

378 citations


Journal ArticleDOI
TL;DR: A novel iterative algorithm is proposed to optimize the waveforms and receiving filters in the MIMO radar such that the detection performance can be maximized and these algorithms have better SINR performance than existing design methods.
Abstract: The concept of multiple-input multiple-output (MIMO) radar allows each transmitting antenna element to transmit an arbitrary waveform. This provides extra degrees of freedom compared to the traditional transmit beamforming approach. It has been shown in the recent literature that MIMO radar systems have many advantages. In this paper, we consider the joint optimization of waveforms and receiving filters in the MIMO radar for the case of extended target in clutter. A novel iterative algorithm is proposed to optimize the waveforms and receiving filters such that the detection performance can be maximized. The corresponding iterative algorithms are also developed for the case where only the statistics or the uncertainty set of the target impulse response is available. These algorithms guarantee that the SINR performance improves in each iteration step. Numerical results show that the proposed methods have better SINR performance than existing design methods.

275 citations


Journal ArticleDOI
TL;DR: A stochastic non-line-of-sight (NLOS) ultraviolet (UV) communication channel model is developed using a Monte Carlo simulation method based on photon tracing, which captures the multiple scattering effects of UV signal propagation in the atmosphere, and relaxes the assumptions of single scattering theory.
Abstract: A stochastic non-line-of-sight (NLOS) ultraviolet (UV) communication channel model is developed using a Monte Carlo simulation method based on photon tracing. The expected channel impulse response is obtained by computing photon arrival probabilities and associated propagation delay at the receiver. This method captures the multiple scattering effects of UV signal propagation in the atmosphere, and relaxes the assumptions of single scattering theory. The proposed model has a clear advantage in reliable prediction of NLOS path loss, as validated by outdoor experiments at small to medium elevation angles. A Gamma function is shown to agree well with the predicted impulse response, and this provides a simple means to determine the channel bandwidth. The developed model is employed to study the characteristics of NLOS UV scattering channels, including path loss and channel bandwidth, for a variety of scattering conditions, source wavelength, transmitter and receiver optical pointing geometries, and range.

256 citations


Journal ArticleDOI
TL;DR: A method for eliciting a novel ERP, known as the AESPA (auditory-evoked spread spectrum analysis), which uses rapid amplitude modulation of audio carrier signals to estimate the impulse response of the auditory system.
Abstract: In natural environments complex and continuous auditory stimulation is virtually ubiquitous. The human auditory system has evolved to efficiently process an infinitude of everyday sounds, which range from short, simple bursts of noise to signals with a much higher order of information such as speech. Investigation of temporal processing in this system using the event-related potential (ERP) technique has led to great advances in our knowledge. However, this method is restricted by the need to present simple, discrete, repeated stimuli to obtain a useful response. Alternatively the continuous auditory steady-state response is used, although this method reduces the evoked response to its fundamental frequency component at the expense of useful information on the timing of response transmission through the auditory system. In this report, we describe a method for eliciting a novel ERP, which circumvents these limitations, known as the AESPA (auditory-evoked spread spectrum analysis). This method uses rapid amplitude modulation of audio carrier signals to estimate the impulse response of the auditory system. We show AESPA responses with high signal-to-noise ratios obtained using two types of carrier wave: a 1-kHz tone and broadband noise. To characterize these responses, they are compared with auditory-evoked potentials elicited using standard techniques. A number of similarities and differences between the responses are noted and these are discussed in light of the differing stimulation and analysis methods used. Data are presented that demonstrate the generalizability of the AESPA method and a number of applications are proposed.

203 citations


Patent
21 Dec 2009
TL;DR: In this article, a distributed antenna system consisting of a signal path within a signal processing board, a compensator coupled to the signal path, and a finite impulse response (FIR) filter with an impulse response function represented by H(ω), the compensator having an FIR filter parameter vector {right arrow over (h)} determined from an estimated system response y(n) of the signalpath to an input comb signal x(n), wherein y n is estimated from interpolated measured output responses from a plurality of frequency sweep signal test inputs.
Abstract: Systems and methods for digitally equalizing a signal in a distributed antenna system are provided. In one embodiment, a signal processing device within a distributed antenna system comprises a signal path within a signal processing board, the signal path having an uncompensated distortion function of G(ω) with a system response represented by y(n); and a compensator coupled to the signal path, the compensator having a finite impulse response (FIR) filter with an impulse response function represented by H(ω), the compensator having an FIR filter parameter vector {right arrow over (h)} determined from an estimated system response y(n) of the signal path to an input comb signal x(n), wherein y(n) is estimated from interpolated measured output responses of the signal path to a plurality of frequency sweep signal test inputs.

203 citations


Journal ArticleDOI
TL;DR: In this paper, the authors developed a field-coupling model for propagation in graded-index MMF, analogous to the principal-state model for polarization-mode dispersion in single-mode fiber.
Abstract: Power-coupling models are inherently unable to describe certain mode coupling effects in multimode fiber (MMF) when using coherent sources at high bit rates, such as polarization dependence of the impulse response. We develop a field-coupling model for propagation in graded-index MMF, analogous to the principal-states model for polarization-mode dispersion in single-mode fiber. Our model allows computation of the fiber impulse response, given a launched electric-field profile and polarization. In order to model both spatial- and polarization-mode coupling, we divide a MMF into numerous short sections, each having random curvature and random angular orientation. The model can be described using only a few parameters, including fiber length, number of sections, and curvature variance. For each random realization of a MMF, we compute a propagation matrix, the principal modes (PMs), and corresponding group delays (GDs). When the curvature variance and fiber length are small (low-coupling regime), the GDs are close to their uncoupled values, and scale linearly with fiber length, while the PMs remain highly polarized. In this regime, our model reproduces the polarization dependence of the impulse response that is observed in silica MMF. When the curvature variance and fiber length are sufficiently large (high-coupling regime), the GD spread is reduced, and the GDs scale with the square root of the fiber length, while the PMs become depolarized. In this regime, our model is consistent with the reduced GD spread observed in plastic MMF.

180 citations


Journal ArticleDOI
TL;DR: This work describes three sampling regimes for FFT-based propagation approaches: ideally sampled, oversampled, and undersampled and describes the form of the sampled chirp functions and their discrete transforms.
Abstract: Accurate simulation of scalar optical diffraction requires consideration of the sampling requirement for the phase chirp function that appears in the Fresnel diffraction expression. We describe three sampling regimes for FFT-based propagation approaches: ideally sampled, oversampled, and undersampled. Ideal sampling, where the chirp and its FFT both have values that match analytic chirp expressions, usually provides the most accurate results but can be difficult to realize in practical simulations. Under- or oversampling leads to a reduction in the available source plane support size, the available source bandwidth, or the available observation support size, depending on the approach and simulation scenario. We discuss three Fresnel propagation approaches: the impulse response/transfer function (angular spectrum) method, the single FFT (direct) method, and the two-step method. With illustrations and simulation examples we show the form of the sampled chirp functions and their discrete transforms, common relationships between the three methods under ideal sampling conditions, and define conditions and consequences to be considered when using nonideal sampling. The analysis is extended to describe the sampling limitations for the more exact Rayleigh-Sommerfeld diffraction solution.

122 citations


Journal ArticleDOI
TL;DR: An unbiased RTF estimator is developed that exploits the nonstationarity and presence probability of the speech signal and derive an analytic expression for the estimator variance.
Abstract: In this paper, we present a relative transfer function (RTF) identification method for speech sources in reverberant environments. The proposed method is based on the convolutive transfer function (CTF) approximation, which enables to represent a linear convolution in the time domain as a linear convolution in the short-time Fourier transform (STFT) domain. Unlike the restrictive and commonly used multiplicative transfer function (MTF) approximation, which becomes more accurate when the length of a time frame increases relative to the length of the impulse response, the CTF approximation enables representation of long impulse responses using short time frames. We develop an unbiased RTF estimator that exploits the nonstationarity and presence probability of the speech signal and derive an analytic expression for the estimator variance. Experimental results show that the proposed method is advantageous compared to common RTF identification methods in various acoustic environments, especially when identifying long RTFs typical to real rooms.

110 citations


Journal ArticleDOI
TL;DR: In this article, the authors provide a method to construct simultaneous confidence regions for impulse responses and conditional bands to examine significance levels of individual impulse response coefficients given propagation trajectories, and constrain a subset of impulse response paths to anchor structural identification and formally test the validity of such identifying constraints.
Abstract: Inference about an impulse response is a multiple testing problem with serially correlated coefficient estimates. This paper provides a method to construct simultaneous confidence regions for impulse responses and conditional bands to examine significance levels of individual impulse response coefficients given propagation trajectories. The paper also shows how to constrain a subset of impulse response paths to anchor structural identification and how to formally test the validity of such identifying constraints. Simulation and empirical evidence illustrate the new techniques. A broad summary of asymptotic analytic formulas is provided to make the methods easy to implement with commonly available statistical software.

95 citations


Journal ArticleDOI
TL;DR: In this paper, a system identification analysis of a soil-structure interaction model with coupled horizontal and rocking response based on a combination of Fourier analysis, wave travel-time analysis, and a relationship between fixed-base, rigid-body, and system frequencies is presented.
Abstract: This article presents a system identification analysis of a soil-structure interaction model with coupled horizontal and rocking response based on a combination of Fourier analysis, wave travel-time analysis, and a relationship between fixed-base, rigid-body, and system frequencies. The study provides insight into the coupling of the structural and soil vibrations useful for interpretation of seismic recordings in structures. The structural model captures one-dimensional shear-wave propagation in the structure. The analysis shows that the system functions with respect to foundation horizontal motion are those of the coupled soil-structure system, which differs from conclusions of earlier studies based on a model without foundation rocking. The energy of the system vibrational response is concentrated around the frequencies of vibration of the system, which depend on the properties of the structure, soil, and foundation. The analysis shows that the structural fundamental fixed-base (uncoupled) frequency f 1 is related to the wave travel time τ (from the base to the top) by f 1=1/(4 τ ) and that accurate measurement of τ , unaffected by soil-structure interaction, can be obtained from impulse response functions, provided that the data are sufficiently broadband. This is an important result for structural health monitoring because it shows that structural parameters unaffected by soil-structure interaction ( τ , as well as f 1 for structures deforming primarily in shear) can be estimated from seismic monitoring data with minimum instrumentation (two horizontal sensors, one at the base and one at the top). This extends the usability of old strong-motion data in buildings, most of which have not been extensively instrumented, and lessons that can be learned for development and validation of structural health monitoring methodologies. The presented results correspond to a model of the north–south response of the Millikan Library in Pasadena, California, which has become a classical case study for soil-structure interaction.

86 citations


Proceedings ArticleDOI
19 Apr 2009
TL;DR: It is found that MP is efficient and effective to recover CS encoded speech as well as jointly estimate the linear model in the signal dependent unknown linear transform.
Abstract: Compressive sensing (CS) has been proposed for signals with sparsity in a linear transform domain. We explore a signal dependent unknown linear transform, namely the impulse response matrix operating on a sparse excitation, as in the linear model of speech production, for recovering compressive sensed speech. Since the linear transform is signal dependent and unknown, unlike the standard CS formulation, a codebook of transfer functions is proposed in a matching pursuit (MP) framework for CS recovery. It is found that MP is efficient and effective to recover CS encoded speech as well as jointly estimate the linear model. Moderate number of CS measurements and low order sparsity estimate will result in MP converge to the same linear transform as direct VQ of the LP vector derived from the original signal. There is also high positive correlation between signal domain approximation and CS measurement domain approximation for a large variety of speech spectra.

Journal ArticleDOI
TL;DR: In this paper, the authors applied laser pulse shaping techniques to tailor the ultrafast temporal response of localized and propagating optical near fields in resonant optical antennas (ROAs) and plasmonic transmission lines, respectively.
Abstract: We show that laser pulse shaping techniques can be applied to tailor the ultrafast temporal response of localized and propagating optical near fields in resonant optical antennas (ROAs) and plasmonic transmission lines, respectively. Using finite-difference time-domain simulations followed by Fourier transformation, we obtain the impulse response of a nanostructure in the frequency domain, which allows obtaining its temporal response to any arbitrary pulse shape. To illustrate the potential of the method we demonstrate deterministic optimal temporal pulse compression in ROAs with reduced symmetry, in a plasmonic two-wire transmission line, and in a prototype plasmonic circuit combining propagation effects and local resonances. The method described here will be of importance for the coherent control of field propagation in nanophotonic structures and light-induced processes in nanoscopic volumes.

Journal ArticleDOI
TL;DR: In this paper, a high-resolution analysis algorithm and an order-detection method are presented to fill the gap between the low and the high frequency domains (30% < mu < 100%).

Journal ArticleDOI
TL;DR: In this paper, a new technique for rolling element bearing fault diagnosis based on the autocorrelation of wavelet de-noised vibration signal is applied, where the wavelet base function has been derived from the bearing impulse response.
Abstract: Machinery failure diagnosis is an important component of the condition based maintenance (CBM) activities for most engineering systems. Rolling element bearings are the most common cause of rotating machinery failure. The existence of the amplitude modulation and noises in the faulty bearing vibration signal present challenges to effective fault detection method. The wavelet transform has been widely used in signal de-noising, due to its extraordinary time-frequency representation capability. In this paper, a new technique for rolling element bearing fault diagnosis based on the autocorrelation of wavelet de-noised vibration signal is applied. The wavelet base function has been derived from the bearing impulse response. To enhance the fault detection process the wavelet shape parameters (damping factor and center frequency) are optimized based on kurtosis maximization criteria. The results show the effectiveness of the proposed technique in revealing the bearing fault impulses and its periodicity for both simulated and real rolling bearing vibration signals.

Journal ArticleDOI
TL;DR: A reconstruction algorithm based on the convolution formula of diffraction which uses the Fresnel impulse response of free space propagation and makes the use of a numerical spherical wave as a virtual reconstructing wave, thus modifying the virtual reconstruction distance and increasing the kernel bandwidth.
Abstract: This paper presents a reconstruction algorithm based on the convolution formula of diffraction which uses the Fresnel impulse response of free space propagation. The bandwidth of the reconstructing convolution kernel is extended to the one of the object in order to allow the direct reconstruction of objects with size quite larger than the recording area. The spatial bandwidth extension is made possible by the use of a numerical spherical wave as a virtual reconstructing wave, thus modifying the virtual reconstruction distance and increasing the kernel bandwidth. Experimental results confirm the suitability of the proposed method in the case of the simultaneous recording of two-color digital holograms by using a spatial color multiplexing scheme.

Patent
25 May 2009
TL;DR: In this paper, a method for measuring the time of arrival of a signal transmitted from a transmitter (120) to a receiver (110-n) is presented. But the method is limited to the case where the channel impulse response is estimated by combining the received signal portions.
Abstract: Disclosed is a method of measuring time of arrival of a signal transmitted from a transmitter (120) to a receiver (110-n). The method comprises: modulating a plurality of narrowband signal portions onto different carrier frequencies; transmitting, by the transmitter, each modulated signal portion to the receiver; receiving, by the receiver, the transmitted signal portions; estimating the channel impulse response by combining (610) the received signal portions; and measuring (620) the time of arrival using the estimated channel impulse response. Further disclosed is a method of measuring a time of arrival of a signal transmitted from a transmitter to a receiver. The method comprises: estimating a noise level (1310) in an impulse response of a channel between the transmitter and the receiver; finding a first peak (1330) in the channel impulse response that is not noise or a side lobe of a subsequent peak, using the estimated noise level; and measuring the time of arrival (1220) using the first peak.

Journal ArticleDOI
TL;DR: The proposed algorithm has two stages: the first one is similar to the subspace methods as it uses their interpretation as an optimization problem of finding parameters of an optimal multi-step linear predictor for the experimental data, and the second is based on the structured weighted lower rank approximation.

Journal ArticleDOI
TL;DR: In this paper, look-up tables are used to prevent redundant calculations, and a sorting method is used to allow the prevention of unnecessary calculations, which results in a large reduction in computation time.

Patent
Hemanth Sampath1
13 Jul 2009
TL;DR: In this paper, a method of characterizing a frequency response of a transmission channel between a transceiver and a subscriber unit is presented, where the frequency response estimates are converted into a time domain response generating an impulse response once per interval of time.
Abstract: The present invention provides a method of characterizing a frequency response of a transmission channel between a transceiver and a subscriber unit. The method includes once per predetermined interval of time, the transceiver transmitting a signal including multiple carriers, a plurality of the carriers including training symbols, a plurality of the carriers including information symbols. The subscriber unit generates frequency response estimates at the frequencies of the carriers including training symbols, each interval of time. The frequency response estimates are converted into a time domain response generating an impulse response once per interval of time. The impulse responses are filtered over a plurality of intervals of time. A channel profile is determined from the filtered impulse responses. The channel profile is converted to the frequency domain generating a channel interpolator. The characterized frequency response is generated from the channel interpolator and the frequency response estimates. The filtering can include averaging the impulse responses over a plurality of intervals of time, accumulating the impulse responses over a plurality of intervals of time, or weighted averaging of the impulse responses over a plurality of intervals of time. The weighted averaging can be dependent upon a phase error between the impulse responses, and/or an amplitude error between the impulse responses.

Journal ArticleDOI
TL;DR: The lag-augmented vector autoregression method suggested by Toda and Yamamoto (1995) – which models the level of the series but allows for variable inclusion testing on changes in the series – performs well for both Granger causality testing and impulse response function estimation.
Abstract: It is often unclear whether time series displaying substantial persistence should be modelled as a vector autoregression in levels (perhaps with a trend term) or in differences. The impact of this decision on inference is examined here using Monte Carlo simulation. In particular, the size and power of variable inclusion (Granger causality) tests and the coverage of impulse response function confidence intervals are examined for simulated vector autoregression models using a variety of estimation techniques. We conclude that testing should be done using differenced regressors, but that overdifferencing a model yields poor impulse response function confidence interval coverage; modelling in Hodrick-Prescott filtered levels yields poor results in any case. We find that the lag-augmented vector autoregression method suggested by Toda and Yamamoto (1995) – which models the level of the series but allows for variable inclusion testing on changes in the series – performs well for both Granger causality testing and impulse response function estimation.

Proceedings ArticleDOI
30 Sep 2009
TL;DR: A new method to protect the signals from the effects of sparse multipath channels by modulate/encode the signal using random waveforms before transmission and estimate the channel and signal from the observations, without any prior knowledge of the channel other than that it is sparse.
Abstract: Blind deconvolution arises naturally when dealing with finite multipath interference on a signal. In this paper we present a new method to protect the signals from the effects of sparse multipath channels — we modulate/encode the signal using random waveforms before transmission and estimate the channel and signal from the observations, without any prior knowledge of the channel other than that it is sparse. The problem can be articulated as follows. The original message x is encoded with an overdetermined m × n (m > n) matrix A whose entries are randomly chosen; the encoded message is given by Ax. The received signal is the convolution of the encoded message with h, the S-sparse impulse response of the channel. We explore three different schemes to recover the message x and the channel h simultaneously. The first scheme recasts the problem as a block l 1 optimization program. The second scheme imposes a rank-1 structure on the estimated signal. The third scheme uses nuclear norm as a proxy for rank, to recover the x and h. The simulation results are presented to demonstrate the efficiency of the random coding and proposed recovery schemes.

Journal ArticleDOI
TL;DR: The improvement of the calibration technique of null point calorimeters generally used in high enthalpy plasma flows is concerns the accuracy of the identified system in terms of absorbed heat flux during the calibration experiment and of the estimated parameters in the model.
Abstract: This paper concerns the improvement of the calibration technique of null point calorimeters generally used in high enthalpy plasma flows. Based on the linearity assumption, this technique leads to calculate the impulse response that relates the heat flux at the tip of the sensor according to the temperature at the embedded thermocouple close to the heated surface. The noninteger system identification (NISI) procedure is applied. The NISI technique had been well described in previous study. The present work focuses on the accuracy of the identified system in terms of absorbed heat flux during the calibration experiment and of the estimated parameters in the model. The impulse response is thus calculated along with its associated standard deviation. Furthermore, this response is compared with that of the one-dimensional semi-infinite medium, which is classically used in practical applications. The asymptotic behavior of the identified system at the short times is analyzed for a better understanding of the noninteger identified system. Finally, the technique was applied to a new sensor geometry that has been developed particularly for high enthalpy plasma flows and it is shown that the method can be applied to any geometry suitable for a certain test configuration.

Journal ArticleDOI
TL;DR: In this paper, a fast methodology that employs only two anti-polarity one-bit data patterns instead of the pseudo-random bit sequence as input sources to simulate the worst-case eye diagram was proposed.
Abstract: As the speed of signal through an interconnection increases toward the multigigabit ranges, the effects of lossy transmission lines on the signal quality of printed circuit boards becomes a critical issue. To evaluate the eye diagram and thus the signal integrity in the modern digital systems, this paper proposes a fast methodology that employs only two anti-polarity one-bit data patterns instead of the pseudo-random bit sequence as input sources to simulate the worst-case eye diagram. Analytic expressions are derived for the impulse response of the lossy transmission lines due to the skin-effect loss, while the Kramers-Kronig relations are employed to deal with the noncausal problem related to the dielectric loss. Two design graphs that can be used to rapidly predict the eye diagram characteristics versus the conductive and dielectric losses are then constructed and based on which, the maximally usable length of transmission lines under a certain signal specification can be easily acquired. At last, the time-domain simulations and experiments are implemented to verify the exactitude of proposed concept.

Journal ArticleDOI
TL;DR: It is shown that it is possible to restore in vivo ultrasound images using an assumed point-spread function and hence it is concluded that an exact point- spread function is not necessary for enhancing ultrasound image quality by deconvolution.

PatentDOI
TL;DR: In this article, a scheme to design an audio precompensation controller for a multichannel audio system, with a prescribed number N of loudspeakers in prescribed positions so that listeners positioned in any of P>1 spatially extended listening regions should be given the illusion of being in another acoustic environment that has L sound sources located at prescribed positions in a prescribed room acoustics.
Abstract: A scheme to design an audio precompensation controller for a multichannel audio system, with a prescribed number N of loudspeakers in prescribed positions so that listeners positioned in any of P>1 spatially extended listening regions should be given the illusion of being in another acoustic environment that has L sound sources located at prescribed positions in a prescribed room acoustics. The method provides a unified joint solution to the problems of equalizer design, crossover design, delay and level calibration, sum-response optimization and up-mixing. A multi-input multi-output audio precompensation controller is designed for an associated sound generating system including a limited number of loudspeaker inputs for emulating a number of virtual sound sources. Method includes: estimating, for each loudspeaker input signals, an impulse response at each of a set of measurement positions that cover the P listening regions; specifying a target impulse response (target stages) for each virtual sound source at each measurement position; and determining adjustable filter parameters of the audio precompensation controller so that a criterion function is optimized.

Journal ArticleDOI
TL;DR: A method to adaptively determine the filter length, based on estimation of the mean square deviation, is proposed, primarily intended for identifying long non-sparse systems, such as a typical impulse response from an acoustic setup.

Journal ArticleDOI
TL;DR: In this paper, the frequency-dependent excitation signals for each transducer were calculated using a discretely measured target radiation pattern to calculate the frequency dependent excitation signal for each source.
Abstract: Sound sources for measurements in room acoustics are of omni-directional type, in general. With respect to auralization applications, an omni-directionally measured room impulse response may not be the ideal choice since it does not represent the real life situation playing an instrument in the room. To achieve the directivity of a real source (like a musical instrument or human voice) with a technical sound source (like a loudspeaker) requires either to copy the entire body and the sound radiation (i.e. the surface velocity) of that particular source or to reproduce the directional pattern of the radiation using a multiple source configuration like a dodecahedron or icosahedron loudspeaker array with independent excitation of each transducer. The advantage of the latter method is obvious since one single source is able to provide a large variety of different directivities by simply changing the excitation profile. To maintain the appropriate excitation of each individual transducer, different approaches can be made. In this paper a method is described using a discretely measured target radiation pattern to calculate the frequency dependent excitation signals for each transducer. Hereby, the directivity pattern of the source transducers and a phase optimization of the energy averaged radiation of musical instruments are used. The advantage of this method is a very flexible computation that is able to match the radiation pattern at the points of measurement very well. The resulting input filters for the platonic sound source can be used either for a real time convolution or offline processing of measured signals.

Patent
09 Apr 2009
TL;DR: In this article, an apparatus for generating filter characteristics for filters connectible to at least three loudspeakers at defined locations with respect to a sound reproduction zone comprises an impulse response reverser (10) for time-reversing impulse responses associated to the loudspeakers to obtain timereversed impulse responses.
Abstract: An apparatus for generating filter characteristics for filters connectible to at least three loudspeakers at defined locations with respect to a sound reproduction zone comprises an impulse response reverser (10) for time-reversing impulse responses associated to the loudspeakers to obtain time-reversed impulse responses. The apparatus furthermore comprises an impulse response modifier (14) for modifying the impulse responses or the time-reversed impulse responses such that impulse response portions occurring before a maximum of a time-reversed impulse response are reduced in amplitude to obtain the filter characteristics for the filters.

01 Mar 2009
TL;DR: A novel theoretical framework is presented for using recent advances in frequency diversity arrays (FDAs) and an innovative solution for using the convolution back-projection algorithm, the gold standard in SAR image processing, is a significant advantage of the proposed FDA model.
Abstract: : In this work, a novel theoretical framework is presented for using recent advances in frequency diversity arrays (FDAs). Unlike a conventional array, the FDA simultaneously transmits a unique frequency from each element in the array. As a result, special time and space properties of the radiation pattern are exploited to improve cross-range resolution. The idealized FDA radiation pattern is compared with and validated against a full-wave electromagnetic solver, and it is shown that the conventional array is a special case of the FDA. A new signal model, based on the FDA, is used to simulate SAR imagery of ideal point mass targets and the new model is used to derive the impulse response function of the SAR system, which is rarely achievable with other analytic methods. This work also presents an innovative solution for using the convolution back-projection algorithm, the gold standard in SAR image processing, and is a significant advantage of the proposed FDA model. The new FDA model and novel SAR system concept of operation are shown to reduce collection time by 33 percent while achieving a 4.5 dB improvement in cross-range resolution as compared to traditional imaging systems.

Proceedings ArticleDOI
01 Dec 2009
TL;DR: This work relates the Cramer-Rao bounds on eigenvalue estimator performance to structural properties of the transfer function, and in turn to the network's topological structure, and finds that stimulus and observation in each strongly-connected network component is needed for high-fidelity estimation.
Abstract: We examine the role played by a linear dynamical network's topology in inference of its eigenvalues from noisy impulse-response data. Specifically, for a canonical linear time-invariant network dynamics, we relate the Cramer-Rao bounds on eigenvalue estimator performance (from impulse-response data) to structural properties of the transfer function, and in turn to the network's topological structure. We focus especially on networks with a slow-coherence structure, in which case we find that stimulus and observation in each strongly-connected network component is needed for high-fidelity estimation.