Abstract: A sound processing method and apparatus are provided, which are capable of performing sound processing on input audio signals containing a plurality of signal components being different in desired sound processing conditions, in a manner that allows natural sound to be reproduced. An input audio signal of at least one system is separated into a plurality of separated signal components, and each signal component of at least part of the plurality of separated signal components is subjected to individual sound processing according to the signal component, and the plurality of separated signal components are outputted as at least one audio signal after each signal component of the at least part thereof is subjected to the individual sound processing. The plurality of separated signal components are synthesized into a synthesized audio signal, which is then outputted, or alternatively, the plurality of separated signal components are outputted separately as audio signals.

Abstract: In the noise suppression apparatus, a spectrum correction gain calculation unit calculates the noise amplitude spectrum correction gain and the noise removal spectrum correction gain using the input amplitude spectrum, noise amplitude spectrum and respective coefficients; a spectrum deduction unit deducts the product of the noise amplitude spectrum and the noise amplitude spectrum correction gain from the input amplitude spectrum and outputs the result as a first noise removal spectrum; a spectrum suppression unit multiplies the first noise removal spectrum by the noise removal spectrum correction gain and outputs the result as a second noise removal spectrum; finally a frequency/time conversion unit converts the second noise removal spectrum into a time domain signal.

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: There is provided an adjustable harmonic distortion detector that includes a clock signal source, means for the detection of a first period of evaluation, and means for the detection of a second period of evaluation. The detector has the characteristic that a first block memorizes a number equal to the clock pulses present in the first period of evaluation, a multiplier block performs a multiplication between the number stored in the first block and a multiplicative factor during the second period of evaluation, and a second block memorizes the outcome. The second block is adapted to generate an output signal when the outcome in the second block is equal to zero.

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 is provided of extracting desired signals st from contaminated signals yt measured via respective communication channels. The system comprising the desired signals st and the channels is modelled as a state space model. In the model, the desired signals have time-varying characteristics which vary more quickly than second time-varying characteristics of the channels.

Abstract: A system and method facilitating signal enhancement utilizing mixture models is provided. The invention includes a signal enhancement adaptive system having a speech model, a noise model and a plurality of adaptive filter parameters. The signal enhancement adaptive system employs probabilistic modeling to perform signal enhancement of a plurality of windowed frequency transformed input signals received, for example, for an array of microphones. The signal enhancement adaptive system incorporates information about the statistical structure of speech signals. The signal enhancement adaptive system can be embedded in an overall enhancement system which also includes components of signal windowing and frequency transformation.

Abstract: A self-adaptive graphic equalizer operable to equalize the affects of an audio system on an audio signal includes an adaptive graphic equalizer having a plurality of equalizing filters, where the plurality of equalizing filters have different center frequencies equidistant from one another and spanning a predetermined audio bandwidth. Each equalizing filter is operable to filter an ith sub-band of the audio signal. A plurality of first filters are coupled to the audio system, each first filter is operable to filter an ith sub-band of an output signal of the audio system. A plurality of second filters are operable to filter an ith sub-band of the audio signal. A gain adjuster is operable to adjust the ith sub-band of the adaptive graphic equalizer in response to a difference in the ith sub-band of the filtered output signal from the plurality of first filters and the ith sub-band of the filtered audio signal from the plurality of second filters.

Abstract: A method and apparatus for de-noising weak bio-signals having a relatively low signal to noise ratio utilizes an iterative process of de-noising a data set comprised of a new set of frames. The method separately performs a non-linear de-noising operation on each of the component frames and combines the resultant de-noised frames to form a combined resultant de-noised input signal. The method is preferably carried out in a digital processor.

Abstract: A method and apparatus for de-noising weak bio-signals having a relatively low signal to noise ratio utilizes an iterative process of wavelet de-noising a data set comprised of a new set of frames of wavelet coefficients partially generated through a cyclic shift algorithm. The method preferably operates on a data set having 2N frames, and the iteration is performed N?1 times. The resultant wavelet coefficients are then linearly averaged and an inverse discrete wavelet transform is performed to arrive at the de-noised original signal. The method is preferably carried out in a digital processor.

Abstract: Sound source separation, without permutation, using convolutional mixing independent component analysis based on a priori knowledge of the target sound source is disclosed. The target sound source can be a human speaker. The reconstruction filters used in the sound source separation take into account the a priori knowledge of the target sound source, such as an estimate the spectra of the target sound source. The filters may be generally constructed based on a speech recognition system. Matching the words of the dictionary of the speech recognition system to a reconstructed signal indicates whether proper separation has occurred. More specifically, the filters may be constructed based on a vector quantization codebook of vectors representing typical sound source patterns. Matching the vectors of the codebook to a reconstructed signal indicates whether proper separation has occurred. The vectors may be linear prediction vectors, among others.

Abstract: A noise suppressor device for attaining perceptually preferable noise suppression is disclosed. The device minimizes reduction in quality even in the presence of increased noises. The device is adaptable for use in voice communications systems and speech recognition systems employed in a variety of kinds of noisy environments. The device includes a spectrum subtracter and a spectrum amplitude suppressor that operate on the basis of perceptual weights.

Abstract: A method and apparatus for speaker recognition is provided that matches the noise in training data to noise in testing data using spectral addition. Under spectral addition, the mean and variance for a plurality of frequency components are adjusted in the training data and the test data so that each mean and variance is matched in a resulting matched training signal and matched test signal. The adjustments made to the training data and test data add to the mean and variance of the training data and test data instead of subtracting from the mean and variance.

Abstract: Disclosed is an apparatus for and a method of filtering noise from a mixed sound signal to obtained a filtered target signal, comprising the steps of inputting the mixed signal through a pair of microphones into a first channel and a second channel, separately Fourier transforming each said mixed signal into the frequency domain, computing a signal short-time spectral amplitude |?| from said transformed signals, computing a signal short-time spectral complex exponential ei arg(S) from said transformed signals, where arg(S) is the phase of the target signal in the frequency domain, computing said target signal S in the frequency domain from said spectral amplitude and said complex exponential.

Abstract: A machine model estimating device of an electric motor control apparatus having an electric motor for driving a load machine, a rotation detector for detecting a rotating angle of the electric motor, and a servo control device for controlling the electric motor. The machine has a calculating device for outputting an operation command signal for operating the electric motor to the servo control device, and frequency characteristic equations for a rigid body model and an N-inertia model, N being an integer which is equal to or greater than 2, which are previously input to the calculating device. The calculating device includes a frequency characteristic measuring section, a frequency characteristic peak detecting section, an attenuation estimation value analyzing section, a frequency characteristic error calculating section, and a machine model deciding section.

Abstract: Sound source separation, without permutation, using convolutional mixing independent component analysis based on a priori knowledge of the target sound source is disclosed. The target sound source can be a human speaker. The reconstruction filters used in the sound source separation take into account the a priori knowledge of the target sound source, such as an estimate the spectra of the target sound source. The filters may be generally constructed based on a speech recognition system. Matching the words of the dictionary of the speech recognition system to a reconstructed signal indicates whether proper separation has occurred. More specifically, the filters may be constructed based on a vector quantization codebook of vectors representing typical sound source patterns. Matching the vectors of the codebook to a reconstructed signal indicates whether proper separation has occurred. The vectors may be linear prediction vectors, among others.

Abstract: The invention relates to a process and a system for voice recognition in a noisy signal. In a preferred embodiment, the system (2) comprises modules for detecting speech (30) and for formulating a noise model (31), a module (40) for quantifying the energy level of the noise and for comparing with preestablished energy spans, a parameterization pathway (5) comprising an optional denoising module (51), with Wiener filter, a module (52) for calculating the spectral energy in Bark windows, a module (50, 530) for applying a configuration of shift values (531), by adding these values to the Bark coefficients, as a function of the quantification (40), so as to modify the parameterization, a module (54) for calculating vectors of parameters, and a block (6) for recognizing shapes, performing the voice recognition by comparison with vectors of parameters prerecorded during a learning phase.

Abstract: Dividing devices are provided for dividing an input information signal into a plurality of frequency bands, and a level detector is provided for detecting a level of a noise component of the information signal divided by the dividing device. A plurality of thresholds corresponding to the divided information signals are stored in a memory. One or more thresholds are selected from the threshold stored in the memory based on a detected result by the level detector. Attenuators are provided for comparing the information signals with a selected threshold and for attenuating an information signal the level of which is lower than the selected threshold.

Abstract: An input audio signal is analyzed to determine a power spectral density profile and the power spectral density profile is compared with at least one template profile. On the basis of the comparison, frequency bands of the input audio signal are selectively attenuated.

Abstract: A spectral enhancement system is disclosed that includes an input node for receiving an input signal, at least one broad band pass filter coupled to the input node and having a first band pass range, at least one non-linear circuit coupled to the filter for non-linearly mapping a broad band pass filtered signal by a first non-linear factor n, at least one narrow band pass filter coupled to the non-linear circuit and having a second band pass range that is narrower than the first band pass range, and an output node coupled to the narrow band pass filter for providing an output signal that is spectrally enhanced.

Abstract: A method for producing an approximated partial transfer function can be used in an electroacoustic appliance for producing an environment correction transfer function that matches an appliance transfer function for the electroacoustic appliance to an acoustic environment, by a) providing a number of basic functions, which each have one basic characteristic of a spectral profile of partial transfer functions, b) providing the approximated partial transfer function by combination of the basic functions weighted by weighting factors, in that at the weighting factor is in each case determined for each basic function such that operation of the electroacoustic appliance is matched to an acoustic environment taking into account the approximated partial transfer function which is formed by the weighting factors and the basic functions, and c) storing the approximated partial transfer function in the electroacoustic appliance for use during operation.

Abstract: An adaptive resonance canceller system and method for attenuating narrowband noise signals within an input signal, where the narrowband noise signals may vary significantly in frequency. The sensor comprises a notch filter, an error reference and gradient generator, and a complex correlator circuit for attenuating the narrowband noise component of the input signal. The input signal is applied to the notch filter and to the error reference and gradient generator. The latter component generates an error reference signal and an error gradient signal. These two signals are applied to the complex correlator circuit which, in turn, generates a tuning parameter signal which is applied to the notch filter to tune the filter to the center frequency of the noise component of the input signal.

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: A method and apparatus for shaping quantization noise generated when compressing audio data at a low bit rate is disclosed. A predetermined quantization noise threshold allowed during quantization of sampled audio data and quantization noise energy information of a quantized MDCT coefficient are received in all frequency bands of an audio frequency. The quantization noise energy of the quantized MDCT coefficient is attenuated in a predetermined number of frequency bands in which a difference between the predetermined quantization noise threshold and the quantization noise energy of the quantized MDCT coefficient is large.

Abstract: The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

Abstract: According to one embodiment of the present invention, a digital hearing device is disclosed. The digital hearing aid includes a microphone for receiving sound, which may include an analog signal. The analog signal is converted by a first converter into a digital signal. Filters are provided to divide the digital signal into multiple signal parts. A signal processor may be provided for each signal part, and performs signal processing on its respective signal part. An adder adds the output of the signal processors, which results in a processed digital signal. A second converter converts the processed digital signal back into an analog signal. A speaker then outputs the analog signal. According to another embodiment of the present invention, a method for enhancing sound is provided.

Abstract: There is disclosed a method and apparatus for changing the frequency content of an input spectrum and a method and apparatus for reducing the perceptibility of a component of an input signal. The first aspect involves adjusting frequency components of the input spectrum in response to a time varying adjustment frequency spectrum to produce an output frequency spectrum including adjusted frequency components of the input spectrum. The time varying input spectrum may be produced by selectively addressing a number of individual sub-spectra at different times. In addition, the input spectrum may be divided into a plurality of sub-spectra and each sub-spectrum may be operated on separately by a different adjustment frequency spectrum at different times. In addition, a perceptual model may be used to enhance the adjustment of the input spectrum or sub-spectra.

Abstract: A method and system for inhibiting noise produced by one or more sources of undesired sound from pickup by a speech recognition unit, where a respective transducer is located proximate each source of undesired sound for converting each source of undesired sound to a corresponding electrical signal, and a noise reduction system is coupled to each of the transducers for converting each electrical signal to an equivalent anti-phase electrical signal of equal amplitude. An output of the noise reduction or a signal corresponding thereto is fed to the speech recognition unit so each of the anti-phase electrical signals cancels or reduces a corresponding electrical signal produced by the speech recognition unit upon picking up the undesired sound from the respective source.

Abstract: A first band analyzer divides an acoustic signal received from a sound playback system through an input unit into frequency bands, and generates a first band level. An acoustic signal estimator estimates the band level of the original acoustic signal at the input unit, and generates a second band level for each band. A processor extracts an external noise component which is contained in the acoustic signal using the first band level and the second band level. The external noise can be accurately estimated with less computation than in the related art.

Abstract: A decoding device (100) is a decoding device that generates frequency spectral data from an inputted encoded audio data stream, and includes: a core decoding unit (102) for decoding the inputted encoded data stream and generating lower frequency spectral data representing an audio signal; and an extended decoding unit (104) for generating, based on the lower frequency spectral data, extended frequency spectral data indicating a harmonic structure, which is same as an extension along the frequency axis of the harmonic structure indicated by the lower frequency spectral data, in a frequency region which is not represented by the encoded data stream.

Abstract: An excursion limiter (10) broadly comprises a voltage controlled filter (40) to suppress an audio signal according to a selected threshold at a selected frequency using a control voltage. The voltage controlled filter (40) preferably produces an inversion signal at the frequency, amplifies the inversion signal according to the control voltage to produce a suppression signal, and combines the audio signal with the suppression signal, thus suppressing the audio signal at the frequency and creating a resultant signal. A control voltage generator (42) preferably generates the control voltage using a frequency compensation filter (58), a full-wave rectifier (60), a precision half-wave rectifier (61) and a non-linear compensator (62). The frequency compensation filter (58) isolates an initial component which is rectified by the full-wave rectifier (60), shifted, and rectified again at the precision rectifier (61) before being essentially flattened in a non-linear manner by the non-linear compensator (62).

Abstract: A digital audio signal has a sequence of samples. Extreme values in an audio waveform represented by the digital audio signal are detected. The extreme values include a maximum value and a minimum value. An audio frequency represented by the digital audio signal is detected in response to the number of samples between two temporally-adjacent extreme-corresponding samples. A difference between each extreme value and a value of a sample which immediately precedes the present extreme-corresponding sample is calculated. The calculated differences are multiplied by selected one of predetermined coefficients to get corrective values respectively. A decision is made as to whether or not the detected audio frequency is in one selected from predetermined frequency bands. When the detected audio frequency is in the selected frequency band, a corresponding corrective value is added to the maximum value and a corresponding corrective value is subtracted from the minimum value.

Abstract: A method of normalizing received digital audio data includes decomposing the digital audio data into a plurality of sub-bands and applying a psycho-acoustic model to the digital audio data to generate a plurality of masking thresholds. The method further includes generating a plurality of transformation adjustment parameters based on the masking thresholds and desired transformation parameters and applying the transformation adjustment parameters to the sub-bands to generate transformed sub-bands.

Abstract: A phase-locked loop (PLL) circuit generates a sine wave signal synchronized with an input signal. The sine wave signal is directly supplied to a first multiplier. The sine wave signal is also supplied to a second multiplier after shifting the phase by 90 degrees by an all-pass filter (APF). The first and second multipliers multiply the corresponding input signals by corresponding predetermined gains. An adder sums the products. Sound corresponding to the sum is produced from a speaker to a sound field. A filter-controlling unit controls the gains for the first and second multipliers, respectively, to minimize an error signal “e” for the output level of a microphone installed at a listening position.

Abstract: An object of the present invention is to provide a simple apparatus for and a simple method of embedding and extracting digital information with little clue to a third party as to embedded digital information with less effort, and the embedded information is securely reconstructed thereby. To embed digital information, a band division portion receives a digital image signal, and then divides the same into ten frequency band signals through discrete wavelet transform so as to compute wavelet coefficients. A mapping portion maps inherent digital information to a pseudo-random number string. An information embedding portion embeds the mapped pseudo-random number string in a string structured by every or some of the computed wavelet coefficients in MRR (signals exclusive of an LL3 signal). A band synthesis portion synthesizes the embedded LL3 digital image signal.

Abstract: A sound intelligibility enhancement (SIE) system is disclosed. The SIE system uses a psychoacoustic model and preferably an oversampled filterbank wherein the level of a signal-of-interest that falls below the environmental noise is selectively amplified as a function of the input level and frequency so that it is audible above the noise but never exceeds a predetermined maximum output level as a function of frequency. The SIE system can be combined with active noise cancellation.

Abstract: A method and apparatus for de-noising weak bio-signals having a relatively low signal to noise ratio utilizes an iterative process of wavelet de-noising a data set comprised of a new set of frames of wavelet coefficients partially generated through a cyclic shift algorithm. The method preferably operates on a data set having 2N frames, and the iteration is performed N−1 times. The resultant wavelet coefficients are then linearly averaged and an inverse discrete wavelet transform is performed to arrive at the de-noised original signal. The method is preferably carried out in a digital processor.

Abstract: System (10) is disclosed including an acoustic sensor array (20) coupled to processor (42). System (10) processes inputs from array (20) to extract a desired acoustic signal through the suppression of interfering signals. The extraction/suppression is performed by modifying the array (20) inputs in the frequency domain with weights selected to minimize variance of the resulting output signal while maintaining unity gain of signals received in the direction of the desired acoustic signal. System (10) may be utilized in hearing aids, voice input devices, surveillance devices, and other applications.

Abstract: A system for identifying a model of an acoustic system in the presence of an external noise signal is disclosed. The system includes an acoustic actuator for generating controlled sound within the acoustic system. A sensor receives the controlled sound and the external noise signal and produces a sensed signal. A control system generates a control signal in response to an error signal. The control system includes a system model for generating an estimated response signal. The control system also generates the error signal representing the difference between the sensed signal and the estimated response signal. A masking threshold generator receives the sensed signal and the error signal and produces spectral shaping parameters. A shaped signal generator for receives the spectral shaping parameters and produces a test signal which is provided as an input to the control system.

Abstract: An amplitude suppression quantity denoting a noise suppression level of a current frame is calculated in an amplitude suppression quantity calculating unit (20), a perceptual weight distributing pattern of both a spectral subtraction quantity and a spectral amplitude suppression quantity is determined in a perceptual weight pattern adjusting unit (21), the spectral subtraction quantity and the spectral amplitude suppression quantity given by the perceptual weight distributing pattern are corrected according to a frequency band SN ratio in a perceptual weight correcting unit (7), a noise subtracted spectrum is calculated from an amplitude spectrum, a noise spectrum and a corrected spectral subtraction quantity in a spectrum subtracting unit (8), and a noise suppressed spectrum is calculated from the noise subtracted spectrum and a corrected spectral amplitude suppression quantity in a spectrum suppressing unit (9).

Abstract: A signal processing method divides a first signal of two signals to be compared in similarity into smaller regions, selects one of the regions, and calculates the correlation of the selected one with the other second signal. The method finds a time difference, an expansion factor, and a similarity in one region in which the maximum similarity as the square of the correlation is obtained, and performs integration in the position represented by the time difference and the expansion factor of values based on similarities. The method performs similar processing on all the regions, and evaluates similarity by, in a peak where the integrated value of similarities is a maximum, compares its magnitude with a threshold value. The region corresponding to the peak can be extracted.

Abstract: Disclosed is an apparatus for and a method of filtering noise from a mixed sound signal to obtained a filtered target signal, comprising the steps of inputting (100) the mixed signal through a pair of microphones (10) into a first channel (15a) and a second channel (15b), separately Fourier transforming (110) each said mixed signal into the frequency domain, computing (130) a signal short-time spectral amplitude |Ŝ| from said transformed signals, computing (140) a signal short-time spectral complex exponential ei arg(S) from said transformed signals, where arg(S) is the phase of the target signal in the frequency domain, computing (150) said target signal S in the frequency domain from said spectral amplitude and said complex exponential.

Abstract: A directional signal processing system for beamforming information signals. The system includes an oversampled filterbank, which has an analysis filterbank for transforming the information signals in time domain into channel signals in transform domain, a synthesis filterbank and a signal processor. The signal processor processes the outputs of the analysis filterbank for beamforming the information signals. The synthesis filterbank transforms the outputs of the signal processor to a single information signal in time domain.

Abstract: An input signal is applied to a notch filter having a transfer function that is the inverse of the expected noise signal. The filtered signal is coupled to a first amplifier and the input signal is coupled to a second amplifier. The outputs of the amplifiers are summed. The gains of the amplifiers are oppositely adjusted in response to the magnitude of the input signal. At low amplitude, the filtered signal is amplified more than the unfiltered signal. At high amplitude, the unfiltered signal is amplified more than the filtered signal.

Abstract: Perceptual coding of spatial cues (PCSC) is used to convert two or more input audio signals into a combined audio signal that is embedded with two or more sets of one or more auditory scene parameters, where each set of auditory scene parameters (e.g., one or more spatial cues such as an inter-ear level difference (ILD), inter-ear time difference (ITD), and/or head-related transfer function (HRTF)) corresponds to a different frequency band in the combined audio signal. A PCSC-based receiver is able to extract the auditory scene parameters and apply them to the corresponding frequency bands of the combined audio signal to synthesize an auditory scene. The technique used to embed the auditory scene parameters into the combined signal enables a legacy receiver that is unaware of the embedded auditory scene parameters to play back the combined audio signal in a conventional manner, thereby providing backwards compatibility.

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: Method for eliminating spurious signal components (SS) from an input signal (ES), said method including the characterization, in a signal analysis phase (I), of the spurious signal components (SS) and of the information signal (NS) contained in the input signal (ES), and the determination or generation, in a signal processing phase (II), of the information signal (NS) or estimated information signal (NS′) on the basis of the characterization obtained in the signal analysis phase (I), said characterization of the signal components (SS, NS) being performed under utilization at least of auditory-based features (M1 to Mj).

Abstract: In an apparatus which estimates characteristics of a surrounding noise only when an input signal is soundless and performs a noise reduction or suppression of the input signal based on the estimated result, a signal noise ratio is estimated from the input signal, and an automatic switch or an automatic adjustment is performed so as to execute a noise reduction only when the signal noise ratio is good, otherwise to avoid the noise reduction or make the noise reduction degree smaller.

Abstract: The disclosed method uses the Wiener frequency filtering to suppress noise in noisy sound signals (u(t)). This method includes a preliminary step in which the sound signals (u(t)) to be noise-suppressed are digitized by sampling and subdivided into frames. The method then includes a first series of steps including the creation of a noise model on N frames, the estimating of the spectral density of the noise and of the energy of the noise model and the computing of a coefficient that reflects the statistical dispersion of the noise. It also includes a second series of steps including the computation of the spectral density of the signals to be noise-suppressed fore each frame. The coefficients of the Wiener filter are modified for each successively processed frame, by the parameters determined at the end of the two series of steps, so as to introduce an energy compensation and an adaptive overestimation of the noise.

Abstract: A noise or vibration control system reduces a sampling rate and reduces a control rate to improve computation efficiency. The present invention permits the use of a sample frequency (fs) that is less than twice the frequency of interest (fd). The sensed signals are filtered to extract a particular frequency range with a lower bound given by (2n−1)*fs/2 and an upper bound given by (2n+1)*fs/2, where n is an integer chosen so that the frequency of interest (fd) is within the extracted frequency range. The control commands are also calculated at a reduced rate, which is dependent upon the bandwidth of the tone, rather than the absolute frequency of the tone. Rather than updating the control signals directly on the sampled sensor data yk as it enters the computer, the control computations are done on the harmonic components ak and bk, or equivalently on the magnitude and phase.