V.J. Mathews

University of Utah, Salt Lake City, Utah, United States

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Publications (31)86.95 Total impact

  • Ashutosh Pandey, V. John Mathews
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    ABSTRACT: Digital hearing aids identify acoustic feedback signals and cancel them continuously in a closed loop with an adaptive filter. This scheme facilitates larger hearing aid gain and improves the output sound quality of hearing aids. However, the output sound quality deteriorates as the hearing aid gain is increased. This paper presents two methods to modify the forward path gain in digital hearing aids. The first approach employs a variable, frequency-dependent gain function that is lower at frequencies of the incoming signal where the information is perceptually insignificant. The second method of this paper automatically identifies and suppresses residual acoustical feedback components at frequencies that have the potential to drive the system to instability. The suppressed frequency components are monitored and the suppression is removed when such frequencies no longer pose a threat to drive the hearing aid system into instability. Together, the gain processing methods of this paper provide 8 to 12 dB more hearing aid gain than feedback cancelers with fixed gain functions. Furthermore, experimental results obtained with real world hearing aid gain profiles indicate that the gain processing methods of this paper, individually and combined, provide less distortion in the output sound quality than classical feedback cancelers enabling the use of more comfortable style hearing aids for patients with moderate to profound hearing loss.
    IEEE Transactions on Audio Speech and Language Processing 01/2012; 20:1043-1055. · 1.68 Impact Factor
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    ABSTRACT: This paper describes an adaptive algorithm for selecting perelectrode stimulus intensities and inter-electrode stimulation phasing to achieve desired isometric plantar-flexion forces via asynchronous, intrafascicular multi-electrode stimulation. The algorithm employed a linear model of force production and a gradient descent approach for updating the parameters of the model. The adaptively selected model stimulation parameters were validated in experiments in which stimulation was delivered via a Utah Slanted Electrode Array that was acutely implanted in the sciatic nerve of an anesthetized feline. In simulations and experiments, desired steps in force were evoked, and exhibited short time-to-peak (
    Acoustics, Speech and Signal Processing (ICASSP), 2012 IEEE International Conference on; 01/2012
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    Ultrasound in Obstetrics and Gynecology 10/2011; 38(S1). · 3.56 Impact Factor
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    A. Pandey, V.J. Mathews
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    ABSTRACT: Acoustic feedback limits the gain provided by hearing aids. Digital hearing aids identify acoustic feedback signals and cancel them continuously in a closed loop with an adaptive filter in the digital domain. This scheme facilitates larger hearing aid gain and improves the output sound quality of hearing aids. However, the output sound quality of hearing aids deteriorates as the hearing aid gain is increased. This paper automatically identifies and suppresses residual acoustical feedback components at frequencies that have the potential to drive the system to instability. The suppressed frequency components are monitored and the suppression is removed when such frequencies no longer pose a threat to drive the hearing aid system into instability. Experimental results obtained with real world hearing aid gain profiles using speech and music signals indicate that the method of this paper provides less distortion in the output sound quality than classical feedback cancelers enabling the use of more comfortable style hearing aids for patients with moderate to profound hearing loss.
    Acoustics, Speech and Signal Processing (ICASSP), 2011 IEEE International Conference on; 06/2011
  • Ashutosh Pandey, V John Mathews
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    ABSTRACT: Digital signal processing in modern hearing aids is typically performed in a subband or transform domain that introduces analysis-synthesis delays in the forward path. Long forward-path delays are not desirable because the processed sound combines with the unprocessed sound that arrives at the cochlea through the vent and changes the sound quality. Nonetheless, subband domain processing for digital hearing aids is the most popular choice for hearing aids because of the associated computational simplicity. In this paper, we present an alternative digital hearing aid structure with low-delay characteristics. The central idea in the paper is a low-delay spectral gain shaping method (SGSM) that employs parallel parametric equalization (EQ) filters. The low-delay SGSM provides frequency-dependent amplification for hearing loss compensation with low forward path delays and performs dynamic signal processing such as noise suppression and dynamic range compression. Parameters of the parametric EQ filters and associated gain values are selected using a least-squares approach to obtain the desired spectral response. The low-delay structure also employs an off-the-forward-path, frequency domain adaptive filter to perform acoustic feedback cancellation. Extensive MATLAB simulations and subjective evaluations of the results indicate that the method of this paper is competent with a state-of-the-art digital hearing aid system, but exhibits much smaller forward-path delays.
    IEEE Transactions on Audio Speech and Language Processing 06/2011; · 1.68 Impact Factor
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    ABSTRACT: Although asynchronous intrafascicular multi-electrode stimulation (IFMS) can evoke fatigue-resistant muscle force, a priori determination of the necessary stimulation parameters for precise force production is not possible. This paper presents a proportionally-modulated, multiple-input single-output (MISO) controller that was designed and experimentally validated for real-time, closed-loop force-feedback control of asynchronous IFMS. Experiments were conducted on anesthetized felines with a Utah Slanted Electrode Array implanted in the sciatic nerve, either acutely or chronically ( n = 1 for each). Isometric forces were evoked in plantar-flexor muscles, and target forces consisted of up to 7 min of step, sinusoidal, and more complex time-varying trajectories. The controller was successful in evoking steps in force with time-to-peak of less than 0.45 s, steady-state ripple of less than 7% of the mean steady-state force, and near-zero steady-state error even in the presence of muscle fatigue, but with transient overshoot of near 20%. The controller was also successful in evoking target sinusoidal and complex time-varying force trajectories with amplitude error of less than 0.5 N and time delay of approximately 300 ms. This MISO control strategy can potentially be used to develop closed-loop asynchronous IFMS controllers for a wide variety of multi-electrode stimulation applications to restore lost motor function.
    IEEE transactions on neural systems and rehabilitation engineering: a publication of the IEEE Engineering in Medicine and Biology Society 03/2011; 19(3):325-32. · 2.42 Impact Factor
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    A. Pandey, V.J. Mathews
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    ABSTRACT: Adaptive filters are commonly used to cancel acoustic feedback in hearing aids. The sound quality of hearing aids deteriorates as the hearing aid gain is increased. This paper presents a method that increases gain possible in hearing aids by automatically identifying and suppressing residual acoustical feedback components that have the potential to drive the system to instability. The increase in the hearing aid gain over traditional methods and the output sound quality were evaluated using subjective tests. The results indicate that the method of this paper provides 6 to 8 dB more hearing aid gain and less perceptual distortion in the output sound quality than several competing algorithms to improve the performance of adaptive feedback cancellers.
    Acoustics Speech and Signal Processing (ICASSP), 2010 IEEE International Conference on; 04/2010
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    ABSTRACT: The objectives of this study were to estimate fetal blood pressure non-invasively from two-dimensional color Doppler-derived aortic blood flow and diameter waveforms, and to compare the results with invasively derived human fetal blood pressures available from the literature. Aortic pressures were calculated from digitally recorded color Doppler cineloops of the fetal descending aorta by applying the Womersley model in combination with the two-element Windkessel model, assuming constant pulse wave velocity during the second half of pregnancy. The results were compared with invasively derived human fetal blood pressures obtained from the literature. In 21 normal pregnancies the estimated mean aortic pressure regression line increased linearly from 28 mmHg at 20 weeks of gestation to 45 mmHg at 40 weeks of gestation. The pulse pressure based on the regression line increased linearly from 21 mmHg at 20 weeks of gestation to 29 mmHg at 40 weeks of gestation. The aortic compliance exhibited a log linear relationship with the gestational age and a statistically significant eightfold increase was observed between 20 and 40 weeks. The aortic downstream peripheral resistance exhibited an exponentially decaying relationship across the same gestational age range. Non-invasively derived aortic systolic and diastolic aortic pressures were comparable with previously reported invasively derived systolic and diastolic umbilical arterial pressures; however, the mean pressures differed significantly from those reported in the umbilical artery in a separate study. The aortic systolic pressures calculated in this study were significantly higher than invasively derived left ventricular systolic pressures that have been previously reported in the literature. This study demonstrates the feasibility of estimating arterial blood pressure in the human fetus. The method described is of potential use in assessing fetal blood pressure non-invasively, particularly for studying relative changes with time.
    Ultrasound in Obstetrics and Gynecology 10/2008; 32(5):673-81. · 3.56 Impact Factor
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    A. Pandey, V.J. Mathews, M. Nilsson
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    ABSTRACT: Adaptive filters are commonly used to cancel acoustic feedback in hearing aids. The sound quality of hearing aids deteriorates as the hearing aid gain is increased. This paper presents a method to alter the gain function in digital hearing aids to provide additional amplification and better output sound quality. This approach employs a variable, frequency-dependent gain function that is lower at frequencies of the incoming signal where the information is perceptually insignificant. The increase in stable gain over traditional methods and the output sound quality were evaluated with a psychoacoustic experiment on normal-hearing listeners. The results indicate that the method of this paper provides more hearing aid gain and less distortion in the output sound quality than feedback cancelers with fixed gain functions.
    Acoustics, Speech and Signal Processing, 2008. ICASSP 2008. IEEE International Conference on; 01/2008
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    H.I.K. Rao, V. John Mathews, Young-Cheol Park
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    ABSTRACT: This paper presents a method for jointly designing immersive audio rendering filters for a single listener using loudspeakers. The filters for crosstalk cancellation are assumed to have finite impulse responses and are designed using the minimax criterion. In addition to the traditional Atal-Schroeder crosstalk canceler structure, this paper explores an alternate topology that requires the approximation of a single filter. In general, the minimax approach provides improved low-frequency performance leading to a better overall separation of the direct-path and cross-path transfer functions than least-squares designs. The performance of the single-filter structure is better than that of the traditional crosstalk cancellation structure.
    IEEE Transactions on Audio Speech and Language Processing 12/2007; · 1.68 Impact Factor
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    ABSTRACT: To examine whether the magnitude-squared coherence between uterine and umbilical blood flow velocity waveforms can, in conjunction with estimated fetal weight, uterine and umbilical pulsatility indices, fetal and maternal heart rates, diastolic notching and the amniotic fluid index, create a sensitive and specific model for the prediction of placental dysfunction. Binary logistic prediction models are created for preeclampsia, pregnancy induced hypertension and intrauterine growth restriction in a study group of 284 unselected midtrimester pregnancies. In each study group, the median value of derived parameters were compared with the uncomplicated pregnancy control group. The magnitude-squared coherence function between the uterine and umbilical flow velocity waveforms was found to be a statistically significant predictor of preeclampsia during the midtrimester of pregnancy. The magnitude-squared coherence did not improve the prediction of intrauterine growth restriction or pregnancy induced hypertension. The inclusion of magnitude-squared coherence as one of the prediction parameters may improve the early identification of pregnancies subsequently complicated by preeclampsia.
    Ultrasound in Medicine & Biology 08/2007; 33(7):1057-63. · 2.46 Impact Factor
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    ABSTRACT: This paper presents a method for designing low-delay nonuniform pseudo quadrature mirror filter (QMF) banks. This method is motivated by the work of Li, Nguyen, and Tantaratana, in which the nonuniform filter bank is realized by combining an appropriate number of adjacent sub-bands of a uniform pseudo-QMF bank. In prior work, the prototype filter of the uniform pseudo-QMF bank was constrained to have linear phase and the overall delay associated with the filter bank was often unacceptably large for filter banks with a large number of sub-bands. This paper proposes a pseudo-QMF filter bank design technique that significantly reduces the delay by relaxing the linear phase constraints. An example in which an oversampled critical-band nonuniform filter bank is designed and applied to a two-state modeling speech enhancement system is presented in this paper. Comparison of the performance of this system to competing methods employing tree-structured, linear phase multiresolution analysis indicates that the approach described in this paper strikes a good balance between system performance and low delay
    IEEE Transactions on Signal Processing 06/2007; · 2.81 Impact Factor
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    H.I.K. Rao, V.J. Mathews, Young-Cheol Park
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    ABSTRACT: This paper describes a method for jointly designing the crosstalk cancellation filters to facilitate binaural rendering of audio through loudspeakers. The minimax criterion is used to design the immersive audio rendering filters having finite impulse responses for a single listener using loudspeakers. The work presented is applied to the traditional Atal-Schroeder crosstalk canceler structure. The minimax approach provides improved low frequency performance and a better overall separation of the direct path and cross path transfer functions than the conventional least-squares designs.
    Acoustics, Speech and Signal Processing, 2007. ICASSP 2007. IEEE International Conference on; 05/2007
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    Ultrasound in Obstetrics and Gynecology 01/2007; 30(4):442-442. · 3.56 Impact Factor
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    J. Jeraj, V.J. Mathews
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    ABSTRACT: This paper presents an almost sure mean-square performance analysis of an adaptive Hammerstein filter for the case when the measurement noise in the desired response signal is a martingale difference sequence. The system model consists of a series connection of a memoryless nonlinearity followed by a recursive linear filter. A bound for the long-term time average of the squared a posteriori estimation error of the adaptive filter is derived using a basic set of assumptions on the operating environment. This bound consists of two terms, one of which is proportional to a parameter that depends on the step size sequences of the algorithm and the other that is inversely proportional to the maximum value of the increment process associated with the coefficients of the underlying system. One consequence of this result is that the long-term time average of the squared a posteriori estimation error can be made arbitrarily close to its minimum possible value when the underlying system is time-invariant.
    IEEE Transactions on Signal Processing 07/2006; · 2.81 Impact Factor
  • A. Pandey, V.J. Mathews
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    ABSTRACT: Howling is a significant problem even in digital hearing aids equipped with adaptive feedback cancellation. Among the many causes of howling is the inability of the adaptive filter to track rapid changes in the feedback path. Many systems use howling detectors to detect the start of howling and reduce the hearing aid gain for several seconds to avoid prolonged howling. Unfortunately the inadequate speech pressure levels (SPL) during times when the gain is reduced causes loss of information and reduced intelligibility of speech signals arriving at the patient's ears. This paper presents a new method that switches to a least-squares adaptation scheme with linear complexity at the onset of howling. The method adapts to the altered feedback path quickly and allows the patient to not lose perceivable information. The complexity of the least-squares estimate is reduced by reformulating the least-squares estimate into a Toeplitz system and solving it with a direct Toeplitz solver. In addition, the gain function is changed immediately after howling detection in such a way that the system operates in a stable manner and the distortions caused are not perceived because of temporal masking. Simulation results comparing with a conventional method is presented in the paper to demonstrate the superior howling suppression capabilities of the method
    Acoustics, Speech and Signal Processing, 2006. ICASSP 2006 Proceedings. 2006 IEEE International Conference on; 06/2006
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    H.I.K. Rao, V.J. Mathews, Young-Cheol Park
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    ABSTRACT: This paper presents a novel approach for implementing immersive audio rendering filters for a single listener using loudspeakers. We address the problem of crosstalk cancellation inherent in loudspeaker rendering and propose to implement the crosstalk cancellation filters using minimax finite impulse response (FIR) filters. The formulation is based on the Atal-Schroeder crosstalk canceller. The use of the optimal FIR filter design procedure ensures significant amount of separation between the direct path and the cross path. An alternative topology which requires the approximation of just one filter has also been explored using the same design principles. The minimax techniques provides superior solutions as compared to a least-squares design and the alternate structure is shown to be robust in its performance
    Acoustics, Speech and Signal Processing, 2006. ICASSP 2006 Proceedings. 2006 IEEE International Conference on; 06/2006
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    J. Jeraj, V.J. Mathews
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    ABSTRACT: This paper presents an algorithm that adapts the parameters of a Hammerstein system model. Hammerstein systems are nonlinear systems that contain a static nonlinearity cascaded with a linear system. In this paper, the static nonlinearity is modeled using a polynomial system, and the linear filter that follows the nonlinearity is an infinite-impulse response (IIR) system. The adaptation of the nonlinear components is improved by orthogonalizing the inputs to the coefficients of the polynomial system. The step sizes associated with the recursive components are constrained in such a way as to guarantee bounded-input bounded-output (BIBO) stability of the overall system. This paper also presents experimental results that show that the algorithm performs well in a variety of operating environments, exhibiting stability and global convergence of the algorithm.
    IEEE Transactions on Signal Processing 05/2006; · 2.81 Impact Factor
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    V. John Mathews, Behrouz Farhang-Boroujeny
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    ABSTRACT: ABSTRACT This paper presents a method for designing low-delay nonuniform pseudo QMF banks. The method is motivated by the work of Li, Nguyen and Tantaratana, in which the nonuniform filter bank is realized by combining an appropriate number of adjacent subbands of a uniform pseudo QMF filter bank. In prior work, the prototype filter of the uniform pseudo QMF is constrained to have linear phase and the overall delay associated with the filter bank was often unacceptably large for filter banks with a large number of subbands. By relaxing the linear phase constraints, this paper proposes a pseudo QMF filter bank design technique that significantly reduces the delay. An example that experimentally verifies the capabilities of the design technique is presented.
    01/2006;
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    V. John Mathews, Ying Deng
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    ABSTRACT: ABSTRACT Particle filters have recently been applied to speech enhancement when the input speech signal is modeled as a time-varying autoregressive process with stochastically evolving parameters. This type of modeling results in a nonlinear and conditionally Gaussian statespace system that is not amenable to analytical solutions. Prior work in this area involved signal processing in the fullband domain and assumed white Gaussian noise with known variance. This paper extends such ideas to subband domain particle filters and colored noise. Experimental results indicate that the subband particle filter achieves higher segmental SNR than the fullband algorithm and is effective in dealing with colored noise without increasing the computational complexity.
    01/2006;

Publication Stats

126 Citations
86.95 Total Impact Points

Institutions

  • 2003–2012
    • University of Utah
      • • Department of BioEngineering
      • • Department of Electrical and Computer Engineering
      Salt Lake City, Utah, United States
  • 2005–2008
    • Erasmus MC
      • Department of Obstetrics and Gynaecology
      Rotterdam, South Holland, Netherlands