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Showing papers on "Noise reduction published in 1993"


Proceedings ArticleDOI
31 Oct 1993
TL;DR: An adaptive Gaussian filtering algorithm is proposed in which the filter variance is adapted to both the noise characteristics and the local variance of the signal.
Abstract: Gaussian filtering has been intensively studied in image processing and computer vision. Using a Gaussian filter for noise suppression, the noise is smoothed out, at the same time the signal is also distorted. The use of a Gaussian filter as pre-processing for edge detection will also give rise to edge position displacement, edges vanishing, and phantom edges. Here, the authors first review various techniques for these problems. They then propose an adaptive Gaussian filtering algorithm in which the filter variance is adapted to both the noise characteristics and the local variance of the signal. >

316 citations


Journal ArticleDOI
01 Apr 1993-Chaos
TL;DR: It was found that all proposed methods converge in this ideal case, but not equally fast, and it is suggested that these nonlinear noise reduction schemes should be compared to Wiener-type filters.
Abstract: Recently proposed noise reduction methods for nonlinear chaotic time sequences with additive noise are analyzed and generalized. All these methods have in common that they work iteratively, and that in each step of the iteration the noise is suppressed by requiring locally linear relations among the delay coordinates, i.e., by moving the delay vectors towards some smooth manifold. The different methods can be compared unambiguously in the case of strictly hyperbolic systems corrupted by measurement noise of infinitesimally low level. It was found that all proposed methods converge in this ideal case, but not equally fast. Different problems arise if the system is not hyperbolic, and at higher noise levels. A new scheme which seems to avoid most of these problems is proposed and tested, and seems to give the best noise reduction so far. Moreover, large improvements are possible within the new scheme and the previous schemes if their parameters are not kept fixed during the iteration, and if corrections are included which take into account the curvature of the attracting manifold. Finally, the fact that comparison with simple low‐pass filters tends to overestimate the relative achievements of these nonlinear noise reduction schemes is stressed, and it is suggested that they should be compared to Wiener‐type filters.

264 citations


Journal ArticleDOI
TL;DR: In this paper, the authors demonstrate that the noisy appearance of the elastograms is due to the nonstationary relationship between the pre- and postcompression signals that results in an artifactual modulation of the strain estimates by the amplitude variations of the envelope of the rf signal.

207 citations


Proceedings ArticleDOI
02 Oct 1993
TL;DR: In this article, the results of an extensive study into the production and reduction of acoustic noise and vibration in the switched reluctance drive have been described, and a new control technique has been developed to cancel the stator vibration.
Abstract: This paper describes the results of an extensive study into the production and reduction of acoustic noise and vibration in the switched reluctance drive. Time domain analysis has been used to draw conclusions about the effects of changing the operating parameters of the power electronic controller on the vibration and acoustic noise. Experimental results have been taken from a four-phase switched reluctance drive with one-, two-, or four-phase excitation. The results of this time domain study reveal important information about the stator vibration which would not be apparent from frequency spectra. The results have been used to derive operational concepts for the power electronic controller to reduce the acoustic noise and vibration produced by the drive. A new control technique has been developed to cancel the stator vibration. Using this novel control method, experimental results show that the vibration and acoustic noise produced by a switched reluctance drive can be reduced dramatically without affecting the performance of the drive.

198 citations


PatentDOI
TL;DR: In this paper, a noise reduction circuit for a hearing aid having an adaptive filter for producing a signal which estimates the noise components present in an input signal is presented. But the circuit also includes a signal combiner for combining the input signal with the adjusted noise-estimating signal to produce a noise reduced output signal.
Abstract: A noise reduction circuit for a hearing aid having an adaptive filter for producing a signal which estimates the noise components present in an input signal. The circuit includes a second filter for receiving the noise-estimating signal and modifying it as a function of a user's preference or as a function of an expected noise environment. The circuit also includes a gain control for adjusting the magnitude of the modified noise-estimating signal, thereby allowing for the adjustment of the magnitude of the circuit response. The circuit also includes a signal combiner for combining the input signal with the adjusted noise-estimating signal to produce a noise reduced output signal.

171 citations


PatentDOI
TL;DR: In this paper, a multi-band, digital audio noise filter that is especially useful in the restoration of motion picture film soundtracks is presented, which utilizes a remote fader board having eight faders which permit a user to control thresholds for sixty-four frequency bins.
Abstract: This disclosure provides a multi-band, digital audio noise filter that is especially useful in the restoration of motion picture film soundtracks. More particularly, the preferred embodiment presented herein utilizes a remote fader board having eight faders which permit a user to control thresholds for sixty-four frequency bins. These faders are monitored by a MOTOROLA 56000-series microprocessor, which accepts digitized audio input signals, performs a Fast Fourier Transform upon a 128 sample window to yield signal contribution for each of the sixty-four frequency bins, and derives FIR filter coefficients for noise attenuation. The digitized audio input signals, which have been stored in a circular input buffer, are convolved with the FIR filter and output as the digitized output signals of the restored motion picture soundtrack.

113 citations


Patent
19 May 1993
TL;DR: In this article, a noise reducing method and associated apparatus for use in a medical radiographic imaging system where an image represented by an array of pixels is processed and the processed image is recorded on a recording medium or visualized on a display monitor.
Abstract: A noise reducing method and associated apparatus for use in a medical radiographic imaging system wherein an image represented by an array of pixels is processed and the processed image is recorded on a recording medium or visualized on a display monitor. The processing comprises the steps of a) decomposing an original image into a sequence of detail images or into an array of coefficients representing detail strength at multiple resolution levels and a residual image, b) pixelwise attenuating the detail images or the coefficient arrays according to the locally estimated amount of relevant signal present and in accordance with an estimated noise level, c) reconstructing a processed image by accumulating detail obtained from the attenuated detail images or from the attenuated detail coefficients, and further adding the residual image.

99 citations


Journal ArticleDOI
TL;DR: Stochastic temporal filtering techniques are proposed to enhance clinical fluoroscopy sequences corrupted by quantum mottle and the problem of displacement field estimation is treated in conjunction with the filtering stage to ensure that the temporal correlations are taken along the direction of motion to prevent object blur.
Abstract: Clinical angiography requires hundreds of X-ray images, putting the patients and particularly the medical staff at risk. Dosage reduction involves an inevitable sacrifice in image quality. In this work, the latter problem is addressed by first modeling the signal-dependent, Poisson-distributed noise that arises as a result of this dosage reduction. The commonly utilized noise model for single images is shown to be obtainable from the new model. Stochastic temporal filtering techniques are proposed to enhance clinical fluoroscopy sequences corrupted by quantum mottle. The temporal versions of these filters as developed here are more suitable for filtering image sequences, as correlations along the time axis can be utilized. For these dynamic sequences, the problem of displacement field estimation is treated in conjunction with the filtering stage to ensure that the temporal correlations are taken along the direction of motion to prevent object blur. >

97 citations


PatentDOI
TL;DR: In this paper, a method and system for adaptively reducing noise in frames of digitized audio signals that may include both speech and background noise is presented, where the attenuation applied to the audio frames is modified gradually on a frame-by-frame basis, each sample in a specific frame is attenuated using the value calculated for that frame.
Abstract: A method and system are provided for adaptively reducing noise in frames of digitized audio signals that may include both speech and background noise. Frames of digitized audio signals are processed to determine what attenuation (if any) should be applied to the current frame of digitized audio signals. Initially it is determined whether the current frame of digitized audio signals includes speech information, this determination being based upon an estimate of noise and on a speech threshold value. An attenuation value determined for the previous audio frame is modified based on this determination and applied to the current frame in order to minimize the background noise which thereby improves the quality of received speech. The attenuation applied to the audio frames is modified gradually on a frame-by-frame basis, each sample in a specific frame is attenuated using the value calculated for that frame. The adaptive noise reduction system may be advantageously applied to telecommunication systems in which portable radio transceivers communicate over RF channels because the adaptive noise reduction technique does not significantly increase data processing overhead.

81 citations


Patent
05 Mar 1993
TL;DR: In this article, a motion estimator identifies a motion block within several video frames and determines an approximate velocity vector using a trimmed square estimation procedure, which is applied to each pixel in the motion block based on the velocity vector in order to determine a revised trajectory for the pixels.
Abstract: In a video signal noise reduction system, image pixels are tracked across multiple frames and then averaged to produce respective noise reduced pixel values. A motion estimator identifies a motion block within several video frames and determines an approximate velocity vector using a trimmed square estimation procedure. Trajectory correction is applied to each pixel in the motion block based on the velocity vector in order to determine a revised trajectory for the pixels. This correction is accomplished by determining a difference in position of a motion block between successive video frames. Based upon this revised trajectory, appropriate pixels corresponding to the motion block are obtained. These pixels are used in conjunction with pixel values obtained from each of the processed frames to produce an averaged video frame. Each pixel of the averaged video frame replaces the corresponding pixel in the original frame if the difference between the original pixel and the corresponding averaged pixel is less than the median difference between all of the original and averaged pixels.

78 citations


Journal ArticleDOI
R. Downing1, P. Gebler1, George A. Katopis1
TL;DR: In this paper, the decoupling capacitor efficiency in reducing the power supply differential switching noise of the multichip-module (MCM) package structure employed in the IBM ES/9000 system is described.
Abstract: The experimental procedures and test vehicles used for the characterization of the decoupling capacitor efficiency in reducing the power supply differential switching noise of the multichip-module (MCM) package structure employed in the IBM ES/9000 system are described. The experimental results are summarized for various switching elements. It is demonstrated that careful design of the test vehicles, tester systems, and probes makes the accurate measurement of Delta-1 noise feasible. Experimental results on the BOBCAT tester show that the decoupling capacitor efficiency in reducing the peak of the differential Delta-1 noise is 50-67%. This efficiency can be increased by reducing the effective inductance in the decoupling capacitor current return path. >

PatentDOI
TL;DR: In this paper, a noise-band divider (18) computes spectral components on a sample-by-sample basis, and circuitry for determining the individual gains can therefore update them.
Abstract: A noise-suppression circuit (10) divides the signal from a microphone (12) into a plurality of frequency sub-bands by means of a noise-band divider (18) and a subtraction circuit (36). By means of gain circuits (32) and (34), it applies separate gains to the separate bands and then recombines them in a signal combiner (38) to generate an output signal in which the noise has been suppressed. Separate gains are applied only to the lower subbands in the voice spectrum. Accordingly, the noise-band divider (18) is required to compute spectral components for only those bands. By employing a sliding-discrete-Fourier-transform method, the noise-band divider (18) computes the spectral components on a sample-by-sample basis, and circuitry (50, 52) for determining the individual gains can therefore update them on a sample-by-sample basis, too.

Proceedings ArticleDOI
27 Apr 1993
TL;DR: A novel noise suppression algorithm based on a modified maximum likelihood estimate is developed and a speech detector based on the temporal signal-to-noise ratio information of every subband of a bandpass filter bank is proposed.
Abstract: Frequency-domain noise suppression systems with a single microphone are studied for adverse mobile noise environments. A novel noise suppression algorithm based on a modified maximum likelihood estimate is developed. The algorithm takes into account not only minimum voice distortion but also subjective criteria for the noise naturalness. A speech detector based on the temporal signal-to-noise ratio (SNR) information of every subband of a bandpass filter bank is proposed. The speech detector is proven to be very effective and robust in a rapidly changing noisy background. The proposed noise suppression method in mobile telephone systems demonstrates much better subjective and objective test results than the existing methods. A real-time test shows that 10 dB noise reduction and 9 dB SNR improvements have been achieved even in the most adverse mobile situation. >

PatentDOI
TL;DR: In this paper, a method for active noise reduction based on destructive interference of sound waves in order to reduce the energy in a sound field employs two omnidirectional microphones (M1, M2) provided in connection with a loudspeaker.
Abstract: A method for active noise reduction based on destructive interference of sound waves in order to reduce the energy in a sound field employs two omnidirectional microphones (M1, M2) provided in connection with a loudspeaker. By means of the microphones the acoustic feedback is eliminated by a closed loop consisting of the microphone and the loudspeaker. The loudspeaker used is an open loudspeaker with a dipole characteristic, thus causing one of the microphones to be more sensitive to the far field and thereby to the noise which has to be suppressed. The method is implemented by means of a device which comprises a digital signal processor for processing the microphone signals and which transmits an output signal to the loudspeaker where the feedback component from the loudspeaker is substantially eliminated, while the output signal's phase and amplitude are adjusted in such a manner that an effective cancellation of the noise is obtained in a area around the loudspeaker's near field. The digital signal processor can preferably be implemented in the form of software modules on an integrated circuit. With the method and the device an integrated reduction in noise level of almost 20 dB is achieved depending on how the filtering in the digital signal processor is adapted. In practice a quiet zone can be obtained in the loudspeaker's near field with an attenuation band which extends from approximatively 100-500 Hz.

Journal ArticleDOI
TL;DR: It is demonstrated that it is not difficult to choose the parameters of this algorithm, even though it uses no other information about the underlying dynamics than the data themselves, and the noise reduction is very robust with respect to changes in the choice of parameters.
Abstract: We apply a recently proposed nonlinear noise-reduction method to time sequences from two different experiments. We demonstrate that it is not difficult to choose the parameters of this algorithm, even though we use no other information about the underlying dynamics than the data themselves. The noise reduction is very robust with respect to changes in the choice of parameters. The reliability of the result is tested by an analysis of the corrections. We discuss the effect of noise reduction on estimates of dimensions, entropies, and Liapunov exponents. For comparison we process one of the sets, densely sampled Taylor-Couette flow data, with a global filter based on singular value decomposition.

Journal Article
TL;DR: In this article, the authors apply a recently proposed nonlinear noise-reduction method to time sequences from two different experiments and demonstrate that it is not difficult to choose the parameters of this algorithm, even though they use no other information about the underlying dynamics than the data themselves.
Abstract: We apply a recently proposed nonlinear noise-reduction method to time sequences from two different experiments. We demonstrate that it is not difficult to choose the parameters of this algorithm, even though we use no other information about the underlying dynamics than the data themselves. The noise reduction is very robust with respect to changes in the choice of parameters. The reliability of the result is tested by an analysis of the corrections. We discuss the effect of noise reduction on estimates of dimensions, entropies, and Liapunov exponents. For comparison we process one of the sets, densely sampled Taylor-Couette flow data, with a global filter based on singular value decomposition

Journal ArticleDOI
TL;DR: An analysis of the propagation of spatially uncorrelated as well as spatially correlated additive noise in scale space, when the noise is subjected to fuzzy derivative operators of any order is given.

Journal ArticleDOI
01 Aug 1993
TL;DR: In this paper, an adaptive separable median filter as a postfilter is presented for removing the blocking effects which generally originate from lower-bit-rate image transmission, which adaptively transforms from a traditional median filter to a low-pass filter progressively when the filter is close to the position of blocking effects.
Abstract: A novel adaptive separable median filter as a postfilter is presented for removing the blocking effects which generally originate from lower-bit-rate image transmission. This filter adaptively transforms from a traditional median filter to a low-pass filter progressively when the filter is close to the position of blocking effects. Simulation results demonstrate that this filter is extremely useful not only in removing the blocking effects, but also in maintaining the main advantages of the median filter such as edge preservation and noise reduction. >

PatentDOI
TL;DR: WAM as discussed by the authors is a new method of digitally coding and decoding acoustic signals for data compression and noise reduction, which comprises constructing a filter bank using wavelet transforms of a basic filter impulse function to represent the response of the mammalian cochlea.
Abstract: WAM™ is a new method of digitally coding and decoding acoustic signals for data compression and noise reduction The method comprises constructing a filter bank using wavelet transforms of a basic filter impulse function to represent the response of the mammalian cochlea Data compression is obtained by truncation of a discrete representation Reconstruction relies on the theory of frames and produces a reconstruction method and apparatus based on irregular sampling methods which produces good quality results in a very few stages Actual reconstructions show very good data compression and noise reduction performance

Journal ArticleDOI
TL;DR: The authors have developed a method to reduce noise in 3D phase‐contrast magnetic resonance (MR) velocity measurements by exploiting the property that blood is incompressible and, therefore, the velocity field describing its flow must be divergence‐free.
Abstract: The authors have developed a method to reduce noise in three-dimensional (3D) phase-contrast magnetic resonance (MR) velocity measurements by exploiting the property that blood is incompressible and, therefore, the velocity field describing its flow must be divergence-free. The divergence-free condition is incorporated by a projection operation in Hilbert space. The velocity field obtained with 3D phase-contrast MR imaging is projected onto the space of divergence-free velocity fields. The reduction of noise is achieved because the projection operation eliminates the noise component that is not divergence-free. Signal-to-noise ratio (S/N) gains on the order of 15%-25% were observed. The immediate effect of this noise reduction manifests itself in higher-quality phase-contrast MR angiograms. Alternatively, the S/N gain can be traded for a reduction in imaging time and/or improved spatial resolution.

PatentDOI
TL;DR: In this article, an ignition signal transforming circuit processes an ignition pulse signal to obtain a vibration noise source signal with a frequency spectrum composed of 0.5×n (integers) order components of the engine r.p.m. as the primary source signal.
Abstract: In an automobile compartment noise reduction system, an ignition signal transforming circuit processes an ignition pulse signal to obtain a vibration noise source signal with a frequency spectrum composed of 0.5×n (integers) order components of the engine r.p.m. as the primary source signal. The signal is applied to an adaptive filter and an LMS calculating circuit via a speaker-microphone transmission characteristic correcting circuit. The primary source signal is synthesized by the filter into a cancel signal and then outputted through a speaker as canceling sound. The canceling sound is received by at least one error microphone at a noise receiving point as an error signal. The error signal is applied to the LMS calculating circuit. The LMS circuit updates the filter coefficients of the adaptive filter on the basis of the primary source signal and the error signal so that the error signal can be minimized. The noise reduction system has high reliability with low cost, and is easy to mount.

PatentDOI
TL;DR: In this article, an aircraft engine active noise cancellation system is described, in which the resonant frequency of the noise radiating structure is tuned to permit noise cancellation over a wide range of frequencies.
Abstract: A noise source for an aircraft engine active noise cancellation system in which the resonant frequency of noise radiating structure is tuned to permit noise cancellation over a wide range of frequencies. The resonant frequency of the noise radiating structure is tuned by a plurality of drivers arranged to contact the noise radiating structure. Excitation of the drivers causes expansion or contraction of the drivers, thereby varying the edge loading applied to the noise radiating structure. The drivers are actuated by a controller which receives input of a feedback signal proportional to displacement of the noise radiating element and a signal corresponding to the blade passage frequency of the engine's fan. In response, the controller determines a control signal which is sent to the drivers, causing them to expand or contract. The noise radiating structure may be either the outer shroud of the engine or a ring mounted flush with an inner wall of the shroud or disposed in the interior of the shroud.

Journal ArticleDOI
TL;DR: In this article, the effect of stress-induced longitudinal magnetic orientation on the transition noise properties of thin film media was investigated for uniaxial and spatial uniform compressive stress in the recording direction.
Abstract: Transition noise properties in longitudinal thin film media are studied by micromagnetic modeling. The mechanism for enhancement of transition noise at small bit intervals due to intertransition interaction is investigated. The noise dependence on the medium parameters, such as saturation magnetization and film thickness, is calculated. Reducing either the saturation magnetization or the film thickness yields a reduction of transition noise at low recording densities and a reduction of the noise enhancement at high recording densities. The effect of stress-induced longitudinal magnetic orientation is studied for an application of a uniaxial and spatial uniform compressive stress in the recording direction. The calculation shows that films with higher orientation ratio exhibit higher transition noise at small bit intervals. >

Proceedings ArticleDOI
27 Apr 1993
TL;DR: Preliminary evidence shows that the dual excitation (DE) speech model may be able to improve the intelligibility of noisy speech for hearing impaired listeners.
Abstract: The dual excitation (DE) speech model is applied to the problem of speech enhancement. The use of this model and its novel decomposition of speech into coexisting voiced and unvoiced components allow removal of additive wideband noise from the degraded speech with only the knowledge of the power spectrum of the noise. The unique properties of each component are exploited to improve the performance of the enhancement system. Informal comparisons between the DE speech enhancement system and a traditional spectral subtraction algorithm show a clear preference for the DE enhancement system. Although the amount of noise reduction in the two systems was similar, the DE system did not contain the tonal artifacts which were present in the spectral subtraction system. Preliminary evidence shows that the DE speech enhancement system may be able to improve the intelligibility of noisy speech for hearing impaired listeners. >

Journal ArticleDOI
TL;DR: Computer experiments show by computer experiments that this noise can be significantly reduced by cascading the output of the receiver in the original system into an identical copy of this receiver.
Abstract: The signal recovered from the first reported experimental secure communication system via chaotic synchronization contains some inevitable noise which degrades the fidelity of the original message. In this letter we show by computer experiments that this noise can be significantly reduced by cascading the output of the receiver in the original system into an identical copy of this receiver.

Patent
04 Mar 1993
TL;DR: In this article, a transversal filter is employed to track for non-linearities in system components that manifest themselves as added noise introduced into the signal propagation path by employing cascaded sets of weighting coefficient and scaling factor multiplying stages.
Abstract: The need to employ costly precision components to reduce non-linearities in the signal processing path of noise reduction circuitry such as an echo canceler and decision feedback equalizer is successfully addressed by a transversal filter which is capable of effectively tracking for non-linearities in system components that manifest themselves as added noise introduced into the signal propagation path. This non-linear tracking capability is attained by employing cascaded sets of weighting coefficient and scaling factor multiplying stages. The first set of weighting coefficients effectively modifies the contents of each of the transmitted symbol samples in the transversal filter delay line to produce respective sets of `partial sums` associated with the respective data symbols employed in the data modulation scheme. The second, cascaded set of `scaling` coefficients or factors is employed to scale selected ones of the sets of the partial sums.

Patent
03 Sep 1993
TL;DR: In this paper, a noisy input signal x is filtered with a restoration filter of median type to generate a filtered input signal y. The sum of the absolute differences between filtered and unfiltered signal is calculated for each position of a sliding window within the input signal representing a local estimate of the noise, and is combined with a global measure of input signal noise to compute two coefficients a and b which are respectively applied to the filtered and filtered signal to generate the output signal z=a*x+b*y which is both globally and locally adapted to the structure of displayed images.
Abstract: A noisy input signal x is filtered with a restoration filter of median type to generate a filtered input signal y. The sum of the absolute differences between filtered and unfiltered signal is calculated for each position of a sliding window within the input signal representing a local estimate of the noise, and is combined with a global measure of the input signal noise to compute two coefficients a and b which are respectively applied to the unfiltered and filtered signal to generate the output signal z=a*x+b*y which is both globally and locally adapted to the structure of displayed images. Advantageously different kinds of filters operate in parallel, whereby the kind of filter elected is locally adapted to the picture activity.

Proceedings ArticleDOI
TL;DR: This paper argues that aliasing should be treated as signal-dependent, additive noise, and presents a model-based justification for this argument, and processes (high resolution images of) natural scenes in a way which enables the `aliased component' of the reconstructed image to be isolated unambiguously.
Abstract: We present a model-based argument that, for the purposes of system design and digital image processing, aliasing should be treated as signal-dependent additive noise. By using a computational simulation based on this model, we process (high resolution images of) natural scenes in a way which enables the 'aliased component' of the reconstructed image to be isolated unambiguously. We demonstrate that our model-based argument leads naturally to system design metrics which quantify the extent of aliasing. And, by illustrating several aliased component images, we provide a qualitative assessment of aliasing as noise.

Patent
12 Jul 1993
TL;DR: In this paper, a weighted temporally averaging method was proposed to reduce the amount of motion in parts of successive x-ray images, which reduced both noise breakthrough and the occurrence of trailers.
Abstract: Noise reduction for use in an x-ray examination apparatus is provided for performing weighted temporally averaging in dependence on an amount of motion in parts of successive x-ray images. Further noise reduction is performed by combining temporal averaging with spatial filtering and motion detection. This reduces both noise breakthrough and the occurrence of trailers. In particular, noise breakthrough is appropriately reduced by hi-temporal filtering. Threshold-values for discriminating between noise and motion are computed on the basis of images generated by the x-ray detector.

Journal ArticleDOI
TL;DR: In this paper, an integrated acoustic signal processing system which provides adaptive noise cancellation, acoustic echo cancellation, and adaptive active noise control for hands-free cellular phones is developed, which provides high-quality, full-duplex voice communication, and reduces the acoustic noise inside an automobile.
Abstract: An integrated acoustic signal processing system which provides adaptive noise cancellation, acoustic echo cancellation, and adaptive active noise control for hands-free cellular phones is developed. This system provides high-quality, full-duplex voice communication, and reduces the acoustic noise inside an automobile. The integration of this system with the existing automobile audio system reduces the overall system cost. The musical interference suppression (MIS) process is developed to cancel the interference of the music while updating the coefficients of the adaptive filters. The MIS filters also model the electroacoustics paths on-line for both the acoustic echo canceler and the active noise controller. >