Abstract: A system and method for adaptive estimation and compensation of clock drift in echo cancellers is provided. The invention includes an acoustic echo cancellation system with a built in adaptive clock drift compensation system. The acoustic echo cancellation system has an AEC component that performs acoustic echo cancellation on data from a capture buffer, by also using information derived from a render buffer. The clock drift compensation system has access to this capture buffer and render buffer. The clock drift compensation system includes a clock drift compensator that calculates, based on the current location of the capture data being processed by the AEC component as well as additional information, the ideal location in the render buffer from which the AEC component should process data. The clock drift compensator further adjusts the current location in the render buffer from which the AEC component processes data based, at least in part, upon this ideal location.

Abstract: A transfer function estimating device for estimating a transfer function of a sound, includes: a sound receiving module receiving a sound from a given sound source and converting the sound into a tone signal; a storage module storing first transfer functions of the sound propagating from the given sound source to the sound receiving module and transformation coefficients for converting the first transfer functions into given second transfer functions so as to associate with each other; a reference tone signal acquiring module acquiring a reference tone signal of the sound source; an acquiring module acquiring a transfer function of the sound received by the sound receiving module on the basis of the tone signal and the reference tone signal; a specifying module acquiring a cross-correlation value between the transfer function acquired by the acquiring module and each of the first transfer functions stored in the storage module.

Abstract: In one aspect of the present invention, a method to reduce noise in a noisy speech signal is disclosed The method comprises: applying at least two versions of the noisy speech signal to a first filter, whereby that first filter outputs a speech reference signal and at least one noise reference signal, applying a filtering operation to each of the at least one noise reference signals, and subtracting from the speech reference signal each of the filtered noise reference signals, wherein the filtering operation is performed with filters having filter coefficients determined by taking into account speech leakage contributions in the at least one noise reference signal.

Abstract: As optimal candidate as a control signal (y*) for generating a control sound suppressing noise from a speaker is selected from among a plurality of control signal candidates (y1 to yn) by a selection function unit. For this selection, a residual noise estimation function unit receiving as input a residual noise signal (e) from an error microphone is introduced. The function unit first obtains an estimated value of a noise component using a first transfer characteristic simulating filter. Further, this noise component estimated value and filtered outputs from second transfer characteristic simulating filters are used to obtain residual noise estimated values for the control signal candidates (y1 to yn). Further, the single control signal candidate corresponding to the smallest of these residual noise estimated values is selected and used as the above control signal (y*).

Abstract: An active noise controller can determine the signal transmission characteristics from the power amplifier and the speaker to the microphone without using any special external measuring instrument and calculate a cosine correction value and a sine correction value without using an external computer. The active noise controller uses the cosine correction value and the sine correction value to actively reduce vibrational noise. The measurement mode is selected on touch panel (3), and correction value calculator (22) calculates cosine correction value C0 and sine correction value C1 by using filter coefficients W0 and W1 which allow error signal e?(n) to approach zero. Memory (23) stores these values C0 and C1.

Abstract: In a fan noise canceling system, a feedforward signal generated when a fan speed change occurs and a feedback signal read by a sensor are sent to a signal amplifying unit for signal amplification, and the amplified signals are then sent to a signal converting unit for converting into a digital signal. A hybrid controller receives and corrects the digital signals sent thereto, and conducts rapid convergence algorithm to derive a reverse digital signal. The reverse digital signal is sent back to the signal converting unit for converting into a reverse analog signal, which is then sent to the signal amplifying unit for power amplification and generating a control signal to drive a loudspeaker to produce a reverse acoustic wave, which cancels out wave of fan noise to effectively reduce fan noise without adversely affecting the heat dissipation effect of the fan.

Abstract: In an active noise cancellation system having an adaptive filter that outputs a control signal, first and second speakers that emit a canceling signal generated based on the control signal, a microphone that detects an error signal, a correction filter that corrects the base signal by a correction value to generate a reference signal and a filter coefficient updater that successively updates the adaptive filter coefficient based on the error signal and reference signal such that the error signal is minimized, the correction value of the correction filter is set to a sum obtained by adding the transfer characteristic from the first speaker to the microphone, and a product obtained by multiplying the transfer characteristic from the second speaker to the microphone by the prescribed value, thereby enabling to reduce the number of microphones and avoid the increase in parts, the amount of work to provide complicated wiring to the microphones, and the computational load involved in updating the adaptive filter co

Abstract: A method of building a model for a physical plant in the presence of noise can include initializing the model of the physical plant, wherein the model is characterized by a parameter vector, estimating an output of the model, and computing a composite cost comprising a weighted average of an error between the estimated output from the model and an actual output of the physical plant, and a derivative of the error. The method further can include determining a step size and a model update direction. The model of the physical plant can be updated. The updating step can be dependent upon the step size. Another embodiment can include the steps of determining a Kalman gain and determining an error vector comprised of two entries weighted by a scalar parameter.

Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: In accordance with a particular embodiment of the present invention, a method is offered that includes providing a low-pass filter in an adaptive filter architecture employing a fast steepest descent method. The method further includes decomposing an error signal, injecting a small change to one or more weight parameters for a linear combiner, measuring changes in an error correlation vector, and calculating one or more gradients, wherein the gradients are processed by the low-pass filter. In more particular embodiments, one or more of the gradients are processed by the low-pass-filter in order to remove measurement noise and improve accuracy. In addition, a real gradient of the error correlation vector is monitored such that adaptations can be made due to non-linearity and non-constant characteristics of a channel. The low-pass filter may be replaced with a Kalman Filter for faster convergence.

Abstract: An active noise cancellation system reduces, at a listening position, the power of a noise signal being radiated from a noise source to the listening position. The system includes an adaptive filter that receives a reference signal representing the noise signal, and provides a compensation signal. A bass management unit receives the compensation signal and applies a phase shift to the compensation signal to provide a phase shifted compensation signal. A first acoustic radiator receives the phase shifted compensation signal and radiates audio indicative thereof to the listening position. A second acoustic radiator receives the compensation signal and radiates audio indicative thereof to the listening position. The transfer function characteristics from the input of the bass management system to the listening position approximately matches a desired transfer function.

Abstract: In a method for canceling unwanted signals from at least one external sound source, such as a loudspeaker, by means of headphones provided with microphones, at least known sound signals from the at least one external sound source are compensated by anti-phase sound signals. These sound signals simulate the at least known sound signals from said at least one external sound source in anti-phase. Said anti-phase sound signals are generated in the headphones in response to signals derived from audio input signals of the at least one external sound source in a filter device which is controlled by the resulting microphone signals.

Abstract: A noise cancellation system for a building (10) includes: one or more microphones (22) arranged outside the building (10) to detect external noise; one or more speakers (24) arranged inside the building (10), the speakers (24) selectively producing sound within the building (10); and, a noise cancellation processor (28) that receives an output from the microphones (22) and in response thereto controls the speakers (24) so as to substantially cancel sound propagated into the building (10) due to the external noise detected by the microphones (22).

Abstract: Disclosed herein is a noise canceling system, including: a first sound collection section configured to collect noise and output a first noise signal; a first signal processing section configured to produce a first noise reduction signal for reducing the noise at a predetermined cancel point; a sound emission section configured to emit noise reduction sound based on the first noise reduction signal; a second sound collection section configured to collect noise and output a second noise signal; and a second signal processing section configured to produce a second noise reduction signal for reducing noise at the cancel point.

Abstract: A modified synchronized overlap add (SOLA) algorithm for performing high-quality, low-complexity audio time scale modification (TSM) is described. The algorithm produces good output audio quality with a very low complexity and without producing additional audible distortion during dynamic change of the audio playback speed. The algorithm may achieve complexity reduction by performing the maximization of normalized cross-correlation using decimated signals. By updating the input buffer and the output buffer in a precise sequence with careful checking of the appropriate array bounds, the algorithm may also achieve seamless audio playback during dynamic speed change with a minimal requirement on memory usage.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: In order to suppress as much noise as possible in a hands-free device in a motor vehicle, for example, two microphones (M1, M2) are spaced a certain distance apart, the output signals (MS1, MS2) of which are added in an adder (AD) and subtracted in a subtracter (SU). The sum signal (S) of the adder (AD) undergoes a Fourier transform in a first Fourier transformer (F1), and the difference signal (D) of the subtracter (SU) undergoes a Fourier transform in a second Fourier transformer (F2). From the two Fourier transforms R(f) and D(f), a speech pause detector (P) detects speech pauses, during which a third arithmetic unit (R) calculates the transfer function HT of an adaptive transformation filter (TF). The transfer function of a spectral subtraction filter (SF), at the input of which the Fourier transform R(f) of the sum signal (S) is applied, is generated from the spectral power density Srr of the sum signal (S) and from the interference power density Snn generated by the adaptive transformation filter (TF).

Abstract: A method for noninvasive on-line secondary path modeling for the filtered-X LMS algorithm actively controls periodic noise. The method, based in the frequency domain, uses the concept of linear independence of two equations/two unknowns to arrive at the secondary path estimate. Linear independence of the two equations is achieved by adjusting the control filter output via the filter coefficients prior to the acquisition of the second set of data corresponding to the second equation.

Abstract: The invention relates to a method for canceling feedback in an acoustic system comprising a microphone, a signal path, a speaker and means for detecting presence of feedback between the speaker and the microphone, the method comprising providing a LMS algorithm for processing the signal; where the LMS algorithm operates with a predetermined adaptation speed when feedback is not present; where the LMS algorithm operates an adaptation speed faster than the predetermined adaptation speed when feedback is present, and where the means for detecting the presence of feedback is used to control the adaptation speed selection of the LMS algorithm. The invention further relates to a method comprising using a highpass filter to prevent low-frequency signals from entering the LMS algorithm. The invention still further relates to a method comprising using bandwidth detection means for determining the presence of a feedback signal. The invention also relates to hearing aids for implementing these methods.

Abstract: In order to suppress as much noise as possible in a hands-free device in a motor vehicle, for example, two microphones (M1, M2) are spaced a certain distance apart, the output signals (MS1, MS2) of which are added in an adder (AD) and subtracted in a subtracter (SU). The sum signal (S) of the adder (AD) undergoes a Fourier transform in a first Fourier transformer (F1), and the difference signal (D) of the subtracter (SU) undergoes a Fourier transform in a second Fourier transformer (F2). From the two Fourier transforms R(f) and D(f), a speech pause detector (P) detects speech pauses, during which a third arithmetic unit (R) calculates the transfer function HT of an adaptive transformation filter (TF). The transfer function of a spectral subtraction filter (SF), at the input of which the Fourier transform R(f) of the sum signal (S) is applied, is generated from the spectral power density Srr of the sum signal (S) and from the interference power density Snn generated by the adaptive transformation filter (TF).

Abstract: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

Abstract: A system (400) for reducing non-acoustic noise includes a primary sensor (420), at least one secondary sensor (410), a filter (415), and a summation unit (425). The primary sensor (420) measures pressure and produces a primary pressure signal. The at least one secondary sensor (410) measures pressure and produce a secondary pressure signal. The filter (415) processes the secondary pressure signal to produce a filtered pressure signal. The summation unit (425) subtracts the filtered pressure signal from the primary pressure signal to reduce non-acoustic noise in the primary pressure signal.

Abstract: A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and/or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g., reverberation reducing) based on a non-linear function, for example, the kurtosis metric.

Abstract: A system and method reduces undesired noise or vibration in a vehicle. The ambient vibration is measured and command signals are generated over time. The command signals are generated based upon the measured vibration and based upon a control weighting. By varying the control weighting over time, the maximum possible performance is always obtained subject to the saturation constraints.

Abstract: There are provided a method and an apparatus for reducing random, continuous, non-stationary noise in audio signals, the noisy audio signal being filtered by means of a predetermined filter function. The filter function is determined dynamically having regard to the current properties of the noisy audio signal and/or its constituent parts, and the filter function is also limited dynamically having regard to the current properties of the noise component contained in the noisy audio signal.

Abstract: A system and method for adaptive estimation and compensation of clock drift in echo cancellers is provided. The invention includes an acoustic echo cancellation system with a built in adaptive clock drift compensation system. The acoustic echo cancellation system has an AEC component that performs acoustic echo cancellation on data from a capture buffer, by also using information derived from a render buffer. The clock drift compensation system has access to this capture buffer and render buffer. The clock drift compensation system includes a clock drift compensator that calculates, based on the current location of the capture data being processed by the AEC component as well as additional information, the ideal location in the render buffer from which the AEC component should process data. The clock drift compensator further adjusts the current location in the render buffer from which the AEC component processes data based, at least in part, upon this ideal location.

Abstract: A method for canceling feedback in an acoustic system including a microphone, a signal path, a speaker and means for detecting presence of feedback between the speaker and the microphone, the method including providing an LMS algorithm for processing the signal; where the LMS algorithm operates with a predetermined adaptation speed when feedback is not present; where the LMS algorithm operates an adaptation speed faster than the predetermined adaptation speed when feedback is present, and where the means for detecting the presence of feedback is used to control the adaptation speed selection o the LMS algorithm, where the feedback detection means comprises bandwidth detection means for determining the presence of a feedback signal.

Abstract: The method of noise attenuation comprises the steps of generating a noise canceling signal, sensing for an system condition, and ceasing the generation of the noise canceling based upon the system condition. This method is embodied in a system that includes an air induction body, a speaker in proximity to the air induction body, a sensor for sensing a system condition, and a control unit with a noise cancellation feature. The control unit is in communication with both the speaker and the sensor. Based upon the sensed system condition, the control unit may disable the noise cancellation feature.

Abstract: A method to automatically and adaptively tune a leaky, normalized least-mean-square (LNLMS) algorithm so as to maximize the stability and noise reduction performance in feedforward adaptive noise cancellation systems. The automatic tuning method provides for time-varying tuning parameters ?k and ?k that are functions of the instantaneous measured acoustic noise signal, weight vector length, and measurement noise variance. The method addresses situations in which signal-to-noise ratio varies substantially due to nonstationary noise fields, affecting stability, convergence, and steady-state noise cancellation performance of LMS algorithms. The method has been embodied in the particular context of active noise cancellation in communication headsets. However, the method is generic, in that it is applicable to a wide range of systems subject to nonstationary, i.e., time-varying, noise fields, including sonar, radar, echo cancellation, and telephony.

Abstract: Noise reduction arrangement including: a plurality of actuators (3(n)) for generating secondary noise (ps) to reduce primary noise (pp) and being located in a first surface; a plurality of errors sensors (2(m)) located in a second surface parallel to the first surface for sensing a total amount of noise resulting from the primary noise after being reduced by the secondary noise; a plurality of control elements (5(i)) for controlling the actuators (3(n)) based on the sensor outputs, wherein the distance (d) between the first and second surfaces is such that reduction in power RP of the total amount of noise relative to the primary noise within a predetermined frequency band is within the following range: 0.9×RPmax?RP?RPmax in which RPmax is the maximum obtainable reduction in power of the total amount of noise relative to the primary noise, both RP and RPmax being expressed in decibel.

Abstract: This invention relates to improved methods to generate shaped sweeps, and for the acquisition of vibroseis data with a shaped pilot sweep.

Abstract: An active noise control system is provided to specify an input transducer, an error transducer, an output transducer and an active noise controller for generating an anti-phase canceling acoustic signal to attenuate an input noise and to output a reduced noise. The active noise control system also performs on-line secondary path modeling. For this purpose the active noise control system comprises, in addition to known systems, a secondary path change detection circuitry (6), a signal distinguishing circuitry (7) and an auxiliary noise control circuitry (8), all of them being used to model said secondary path.

Abstract: A method to automatically and adaptively tune a leaky, normalized least-mean-square (LNLMS) algorithm so as to maximize the stability and noise reduction performance in feedforward adaptive noise cancellation systems. The automatic tuning method provides for time-varying tuning parameters &lgr;k and &mgr;k that are functions of the instantaneous measured acoustic noise signal, weight vector length, and measurement noise variance. The method addresses situations in which signal-to-noise ratio varies substantially due to nonstationary noise fields, affecting stability, convergence, and steady-state noise cancellation performance of LMS algorithms. The method has been embodied in the particular context of active noise cancellation in communication headsets. However, the method is generic, in that it is applicable to a wide range of systems subject to nonstationary, i.e., time-varying, noise fields, including sonar, radar, echo cancellation, and telephony.

Abstract: An apparatus and method to minimize acoustic vibration in a vehicle due to impulsive gear clash loads. Sensors are mounted on the vehicle and are used to sense vehicle vibration generated by the gear clash loads. A first gear, having a plurality of teeth and a second gear, also having a plurality of teeth, are disposed such that upon rotation of the first and second gears, certain ones of the second gear teeth mesh with certain ones of the first gear teeth. An actuator, which is coupled to the first gear, drives the first gear and generates torque pulses. A controller is coupled to the actuator and controls the torque pulses that are transmitted to the first gear. The torque pulses generated by the actuator are adjusted in magnitude and phase so that the first gear teeth vibrate at a predetermined frequencies thereby reducing vibrations and peak tooth loading caused by the meshing of the first gear teeth and the second gear teeth.

Abstract: Method for controlling a control arrangement controlling a predetermined system (10), the control arrangement comprising a controller (6), M output sensors (4) providing an output signal vector y(t), L control actuators (2) controlled by a control signal vector u(t) provided by the controller (6) and N reference signal generators (8) for providing a reference signal vector z(t) to the controller (6). The method comprises the step of minimizing a criterion function J defined as a mixture of energy of an observed output signal vector &egr;1(t) and energy of a control error signal vector &egr;2(t), by recursively updating the controller coefficients in w(t) proportional to the observed output signal vector &egr;1 and proportional to the control error signal vector &egr;2.

Abstract: A signal processor for reducing undesirable signal content reduces the undesirable signal content by exaggerating the undesirable signal content and then using this exaggerated undesirable signal and adaptive filter means to estimate the undesirable content in the signal and then substantially removing it from the signal. The signal processor includes a signal mapping means for exaggerating the undesirable signal content; and an adaptive filter means for reducing the undesirable signal content using the exaggerated undesirable signal content.

Abstract: In a hearing aid and a method for processing microphone signals in a hearing aid, a signal processing unit is provided in order to amplify and/or attenuate signal parts of at least two microphone signals in a directionally dependent manner. The hearing aid has a signal analysis unit that is capable of modifying at least one property of the direction-dependent amplification and/or attenuation thereby achieving high transmission quality and noise suppression in a multitude of auditory situations.

Abstract: An adaptive filter suitable for network echo cancellation and other applications contains a coefficient vector update device for feeding coefficient vector updates to a finite impulse response filter in accordance with fast converging algorithms. A double talk detector is included for causing filter adaptation to cease in the presence of double talk in the system being echo cancelled. The coefficient vector update device utilizes a proportional affine projection algorithm to provide fast convergence of the filter system and improved performance over other filter devices utilizing different fast converging algorithms.

Abstract: A method and noise reduction apparatus comprises a microphone array including a plurality of microphone elements for receiving a training signal including a plurality of training signal samples, and a working signal including a plurality of working signal samples, and at least one frequency domain convertor coupled to the plurality of microphone elements for converting the plurality of training signal samples and the plurality of working signal samples to the frequency domain. A signal spatial correlation matrix estimator is coupled to the at least one frequency domain convertor for estimating a signal spatial correlation matrix using the converted plurality of training signal samples. An inverse noise spatial correlation matrix estimator is coupled to the at least one frequency domain convertor for estimating an inverse noise spatial correlation matrix using the converted plurality of working signal samples.

Abstract: A system and method for suppressing cyclic noise components of a sampled sensor signal generates an error compensation signal that is subtracted from the sampled sensor signal to produce a corrected signal. A moving average of the sampled sensor signal, or sensor error signal, is obtained that corresponds to an error signal free of the non-cyclic component. An adaptive noise compensation algorithm is implemented by a compensation module with a minimum number of multiplications. The compensation module relies upon a unit base vector having a length equal to the number of base functions used to represent the cyclic noise component of the sensor signal, or equal to the number of samples over the sampling interval of the cyclic noise. This base vector is multiplied by an adaptive gain, which is a function of the difference between the sensor error signal and a compensation signal output from the adaptive compensation module.

Abstract: A sound pick-up and reproduction system includes at least two sound sensors, such as microphones, situated at different distances from a sound reproduction device, such as a loudspeaker. A processor responds to amplitudes and phases of signals outputted by the at least two sound sensors to control a signal outputted to an amplifier and adapted to drive the sound reproduction device, thereby reducing echoes in the sound reproduction device output caused by acoustic coupling between the at least two sound sensors and the sound reproduction device.

Abstract: A new statistical model describes the corruption of spectral features caused by additive noise. In particular, the model explicitly represents the effect of unknown phase together with the unobserved clean signal and noise. Development of the model has realized three techniques for reducing noise in a noisy signal as a function of the model.

Abstract: A method and system for monitoring, maintaining and analyzing statistical performance information in a signal processing system includes collecting data representing a plurality of aspects of the signal processing system, developing a histogram representative of the data in accordance with a chosen algorithm, and analyzing historical histogram data to generate condition signals for use by the signal processing system.

Abstract: The present invention comprises an adaptive directionality dual microphone system in which the time domain data from the first and second microphones is converted into frequency domain data. The frequency domain data is then manipulated to produce a noise-canceled signal which is converted in an Inverse Fourier Transform block into noise-cancel time domain data.

Abstract: A designing system for adaptively characterizing an audio transmitting system has a white noise generating unit for generating a white noise signal. A speaker radiates the white noise generated by the white noise generating unit into an acoustic space. A microphone is placed at a predetermined position in the acoustic space and collects sound radiated from the speaker. A FIR adaptive filter receives the above white noise signal. An LMS algorithm processing unit updates each tap coefficient of the adaptive filter by using the LMS algorithm. A computation unit calculates the difference between a detection signal output from the microphone and an output of the adaptive filter and outputs the difference as an error signal &egr;. By using a white noise signal having an average power of one, the range of the step size parameter of the LMS algorithm required for stably operating the adaptive filter is fixed.

Abstract: An adaptive signal processing system for improving a quality of a signal. The system includes an analysis filterbank for transforming a primary information signal in time domain into oversampled sub-band primary signals in frequency domain and an analysis filterbank for transforming a reference signal in time domain into oversampled sub-band reference signals. Sub-band processing circuits process the signals output from the filterbanks to improve a quality of an output signal. A synthesis filterbank can combine the outputs of the sub-band processing circuits to generate the output signal.

Abstract: A method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones mi and (p) speakers hpj for the control of said sound-reproducing systems characterized in that:

Abstract: This invention concerns environmental noise monitoring, and more particularly a system and method for environmental noise monitoring. The environmental noise monitoring system comprises three or more sound level transducers arranged apart from each other in an environment containing one or more noise sources. Each transducer is arranged to pick up sounds arriving from all directions and to transduce the sounds into electrical signals. The sound level and direction measuring equipment includes signal samplers connected to respective transducers to receive electrical signals representing environmental sounds and to sample the electrical signals. A computer processor is further provided to perform cross correlation calculations to generate a cross correlation function for each pair of sampled signals.

Abstract: A method to automatically and adaptively tune a leaky, normalized least-mean-square (LNLMS) algorithm so as to maximize the stability and noise reduction performance in feedforward adaptive noise cancellation systems. The automatic tuning method provides for time-varying tuning parameters &lgr;k and &mgr;k that are functions of the instantaneous measured acoustic noise signal, weight vector length, and measurement noise variance. The method addresses situations in which signal-to-noise ratio varies substantially due to nonstationary noise fields, affecting stability, convergence, and steady-state noise cancellation performance of LMS algorithms. The method has been embodied in the particular context of active noise cancellation in communication headsets. However, the method is generic, in that it is applicable to a wide range of systems subject to nonstationary, i.e., time-varying, noise fields, including sonar, radar, echo cancellation, and telephony.

Abstract: A circuit arrangement for controlling audio signal transmissions for a communications system that includes a microphone and a video camera. The arrangement comprises a video processor configured and arranged to receive a video signal from the video camera, detect movement of an object in the video signal, and provide a motion-indicating signal indicating movement relative to the object. An audio processor is coupled to the video processor and is configured and arranged to modify the audio signal to be transmitted responsive to the motion-indicating signal. In another embodiment, a video signal processor is configured and arranged to receive a video signal from the video camera, detect mouth movement of a person and provide a mouth-movement signal indicative of movement of the person's mouth.