Abstract: An information signal reconstructor is configured to reconstruct, using aliasing cancellation, an information signal from a lapped transform representation of the information signal including, for each of consecutive, overlapping regions of the information signal, a transform of a windowed version of the respective region, wherein the information signal reconstructor is configured to reconstruct the information signal at a sample rate which changes at a border between a preceding region and a succeeding region of the information signal.

Abstract: A noise canceling and communication system includes an in-ear device adapted to fit in the ear canal of a device user. A passive noise reduction element reduces external noise entering the ear canal. An external microphone senses an external acoustic signal outside the ear canal. An internal microphone senses an internal acoustic signal proximal to the tympanic membrane. One or more internal sound generators produce a noise cancellation signal and an acoustic communication signal, both directed towards the tympanic membrane. A probe tube shapes an acoustic response between the internal sound generator and the internal microphone to be relatively constant over a wide audio frequency band. An electronics module is located externally of the ear canal and in communication with the in-ear device for processing the microphone signals using a hybrid feed forward and feedback active noise reduction algorithm to produce the noise cancellation signal.

Abstract: An active noise reduction system is provided for receiving an audio input signal and a noise interference signal and calculating an audio broadcasting signal according to a Feedback Filtered-X Least-Mean-Square (FFXLMS) algorithm, wherein the FFXLMS algorithm optimizes a (convergence factor) ? so as to decrease the numbers of divisions operated by the active noise reduction system and increase the operation speed of the active noise reduction system.

Abstract: A system, method, and program product are provided for filtering sound from a selected application on a computer without interrupting voice communications on the computer. The method comprises: monitoring a selected program for an outgoing digital audio signal from a selected application; detecting said digital audio signal; and filtering an analog microphone input with the digital audio signal.

Abstract: A method for determining coefficients of a family of cascaded second order Infinite Impulse Response (IIR) parametric filters used for equalizing a room response. The method includes determining parameters of each IIR parametric filter from poles or roots of a reasonably high-order Linear Predictive Coding (LPC) model. The LPC model is able to accurately model the low-frequency room response modes providing better equalization of loudspeaker and room acoustics, particularly at the low frequencies. Advantages of the method include fast and efficient computation of the LPC model using a Levinson-Durbin recursion to solve the normal equations that arise from the least squares formulation. Due to possible band interactions between the cascaded IIR parametric filters, the method further includes optimizing the Q value of each filter to better equalize the room response.

Abstract: An audio system with at least one audio channel may include a digital audio processor in which at least one digital filter is implemented for each channel. The digital filter of each channel may include: an analysis filter bank configured to receive a broad-band input audio signal and divide the input audio signal into a plurality of sub-bands, a sub-band filter for each sub-band. and a synthesis filter bank configured to receive the filtered sub-band signals and combine them for providing a broad-band output audio signal. A delay is associated with each sub-band signal, the delay of one of the sub-band signals being applied to the broad-band input audio signal upstream of the analysis filter bank and the residual delays being applied to the remaining sub-band signals downstream of the analysis filter bank.

Abstract: Disclosed herein is an echo canceller for use in a sound reinforcement communication system configured to carry out a sound reinforcement communication by utilizing a speaker and a microphone, the echo canceller including: an adaptive filter section configured to adaptively identify an impulse response of a feedback path formed by an acoustic coupling or the like between the speaker and the microphone to estimate an echo component in the feedback path from an input signal to the feedback path, and subtracting the echo component thus estimated from an output signal from the feedback path; and an echo suppressing section configured to execute echo suppressing processing for an output signal from the adaptive filter section.

Abstract: In an aspect, the invention features an active noise reduction device including an electronic signal processing circuit. The electronic signal processing circuit includes a first input for accepting a first signal, a second input for accepting a second signal, an output for providing a third signal, a feed-forward path from the first input to the output, and a feed-forward controller for determining the control parameter by calculating a control signal using the first signal and the second signal and then using the control signal to determine the control parameter. The feed-forward path includes a fixed compensation linear filter and a variable compensation filter having an input for receiving a control parameter that applies a selected linear filter from a family of linear filters that vary in both gain and spectral shape and are selectable by the control parameter.

Abstract: A method and system for active noise cancellation is provided. The system employs subband processing, and preferably implements over-sampled filterbank. The system is applicable to adaptive noise cancellation, adaptive echo cancellation for portable listening devices, such as headsets and other similar listening devices.

Abstract: A noise reduction system and a noise reduction method are provided. The noise reduction system comprises a uni-directional microphone, an omni-directional microphone and a signal processing module. The signal processing module comprises an adaptive noise control (ANC) unit, a main noise reduction unit and an optimizing unit. The uni-directional microphone senses a first audio source to output a first audio signal, and the omni-directional microphone senses a second audio source to output a second audio signal. The ANC unit executes an adaptive noise control to output an estimated signal according to the first audio signal and the second audio signal. The main noise reduction unit executes a main noise reduction process to output a de-noise speech signal according to the estimated signal and the second audio signal. The optimizing unit executes an optimizing process to output an optimized speech signal according to the de-noise speech signal.

Abstract: Methods for modeling the secondary path of an ANC system to improve convergence and tracking during noise control operation, and their associated uses are provided. In one aspect, for example, a method for modeling a secondary path for an active noise control system is provided. Such a method may include receiving a reference signal, filtering the reference signal with an initial secondary path model to obtain a filtered reference signal, calculating an autocorrelation matrix from the filtered reference signal, and calculating a plurality of eigenvalues from the autocorrelation matrix. The method may further include calculating a maximum difference between the plurality of eigenvalues and iterating a test model to determine an optimized secondary path model having a plurality of optimized eigenvalues that have a minimized difference that is less than the maximum difference of the plurality of eigenvalues, such that the optimized secondary path model may be utilized in the active noise control system.

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: To provide a noise control system capable of generating a noise cancellation field where noise is reduced at a desired position in a space. An error scanning filter generates a noise cancellation signal using an adaptive control algorithm based on error scanning, the signal having a phase opposite to that of an acoustic signal component detected by a reference sensor. A control speaker radiates the noise cancellation signal to create a noise cancellation field near the head of a person receiving sound.

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: A system and method is disclosed for harmonizing calibration of audio between a plurality of networked conference rooms to enable each networked conference room to have substantially similar audio characteristics by adjusting speaker gain output of an audio signal sent from a calibration location on a network and tuning microphone response received at a calibration location to a calibrated audio source.

Abstract: The invention relates to a method and a device for passive and active acoustic reduction, including m electro-acoustic bars (41) side by side and separated by gaps (D), thus constituting an open-work acoustic barrier combining passive and active noise-reduction.

Abstract: An active control of an unwanted noise signal at a listening site radiated by a noise source uses a reference signal that has an amplitude and/or frequency such that it is masked for a human listener at the listening site by the unwanted noise signal and/or a wanted signal present at the listening site in order to adapt for the time-varying secondary path in a real time manner such that a user doesn't feel disturbed by an additional artificial noise source.

Abstract: A system of signal processing an input signal in a hearing assistance device to avoid entrainment wherein the hearing assistance device including a receiver and a microphone, the method comprising using an adaptive filter to estimate an acoustic feedback path from the receiver to the microphone, generating one or more estimated future pole positions of a transfer function of the adaptive filter, analyzing stability of the one or more estimated pole positions for an indication of entrainment and adjusting the adaptation of the adaptive filter based on the stability.

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, the adaptive filter modifying at least one of the adaptive coefficients based on a feedback output. The invention further includes a feedback component that provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. The invention further provides a noise statistics component that stores noise statistics associated with a noise portion of an input signal and a signal+noise statistics component that stores signal+noise statistics associated with a signal and noise portion of the input signal.

Abstract: A system and apparatus for constructing a displacement model across a frequency range for a loudspeaker is disclosed. The resultant displacement model is centered around the distortion point. Once a distortion model is constructed it can be incorporated into an audio driver to prevent distortion by incorporating the model and a distortion compensation unit with a conventional audio driver. Various topologies can be used to incorporate a distortion model and distortion compensation unit into an audio driver. Furthermore, a wide variety of distortion compensation techniques can be employed to avoid distortion in such an audio driver.

Abstract: The invention provides a digital circuit arrangement for an ambient noise-reduction system affording a higher degree of noise reduction than has hitherto been possible, through the use of a low latency signal processing chain consisting of analogue-to-digital conversion, digital processing and digital-to-analogue conversion. The arrangement converts the analogue signals into N-bit digital signals at sample rate f0, and then subjects the converted signals to digital filtering. The value of N in some embodiments is 1 but, in any event, is no greater than 8, and f0 may be 64 times the Nyquist sampling rate but, in any event, is substantially greater than the Nyquist sampling rate. This permits digital processing to be used without incurring group delay problems that rule out the use of conventional digital processing in this context.

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: An ANR circuit, possibly of a personal ANR device, reduces the degree of feedforward-based ANR that it provides in response to receiving an indication of the operation of a manually-operable control. The reduction of degree of feedforward-based ANR may be effected by turning off or otherwise deactivating the provision of feedforward-based ANR, reducing a range of frequencies of environmental noise sounds attenuated by the feedforward-based ANR to provide less attenuation of sounds detected by a feedforward microphone that are in a range of frequencies deemed to be those of human speech, and/or creating a notch in the range of frequencies of environmental noise sounds attenuated by the feedforward-based ANR to provide less attenuation of sounds detected by the feedforward microphone that are in a range of frequencies deemed to be those of human speech.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

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 subtractor subtracts an echo canceling signal from a canceling error signal to estimate the resonant noise to be silenced at a position of a microphone, and outputs a first basic signal representing the estimated resonant noise as an input signal supplied to a controller. In the controller, a delay filter generates a second basic signal by delaying the first basic signal by a time value. The controller generates a control signal based on the first basic signal and the second basic signal.

Abstract: A mechanical noise suppression apparatus includes: a framing section adapted to divide an input signal into frames of a predetermined time length; a Fourier transform section adapted to transform framed signals obtained by the framing section into a frequency spectrum of a frequency domain; a mechanical noise reduction section adapted to correct the frequency spectrum of the input signal obtained by the Fourier transform section based on frequency spectrum information of mechanical noise to suppress the mechanical noise; an inverse Fourier transform section adapted to return the frequency spectrum corrected by the mechanical noise reduction section into framed signals of a time domain; and a frame synthesis section adapted to carry out frame synthesis of the framed signals of frames obtained by the inverse Fourier transform section to obtain an output signal in which the mechanical noise is suppressed.

Abstract: When the frequency of an engine rotation signal reaches a predetermined frequency, a comparator of a switching unit outputs a switching control signal to selectors and a filter coefficient updater. Based on the switching control signal, the selector switches from a connection between one memory and a corrector to a connection between another memory and the corrector, thereby changing the transfer characteristics C^rr of the corrector from C^11 to C^10. Based on the switching control signal, the selector switches from a connection between one ADC and a filter coefficient updater to the connection between another ADC and the filter coefficient updater, thereby supplying the filter coefficient updater with an error signal, rather than an error signal.

Abstract: A subtractor subtracts an echo canceling signal ?·y1(n?1) from a canceling error signal e(n) to estimate a residual noise to be silenced at the position of a microphone, and outputs a first basic signal x1(n) representing the residual noise. A first control circuit section generates a first control signal y1(n) based on the first basic signal x1(n) and a second basic signal x2(n) that is generated by delaying the first basic signal x1(n) by a time Z?n. A second circuit section generates a second control signal y2(n) based on the first basic signal x1(n) and an engine rotation signal.

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: A noise extraction method in which an environmental input which includes a noise indicia is selectively modified in accordance with an algorithm that includes one or more factors representing time response, amplitude of response, and error correction. The algorithm may also include thresholding, delay or convergence, among other techniques.

Abstract: Loudspeaker apparatus comprise commonly some sort of casings or assemblies, in which one or more single speakers are integrated and which are used to convert electrical signals into sound. A loudspeaker apparatus 1 for radiating sound in a hemisphere is disclosed having a center axis 9, the loudspeaker apparatus 1 comprising: a set of midrange drivers 6a, b, c, d, e, f, whereby the midrange drivers 6a, b, c, d, e, f are controlled and/or arranged in a Bessel configuration and are operable to provide a first acoustic field in the hemisphere, a set of tweeter drivers 7a, b, c, d adapted to provide a second acoustic field in the hemisphere, whereby the first and the second acoustic field are respectively arranged symmetrically, whereby the main sound emitting directions of the single tweeter drivers 7a, b, c, d are angled to the center axis 9.

Abstract: The loudspeaker device according to the present invention comprises a loudspeaker; a feedforward processing section for performing feedforward processing on an electric signal to be inputted to the loudspeaker based on a preset filter coefficient so that non-linear distortion which occurs from the loudspeaker is removed; and a feedback processing section for detecting vibration of the loudspeaker, and performing feedback processing on an electric signal concerning the vibration with respect to the electric signal to be inputted to the loudspeaker. The feedback processing section performs feedback processing on the electric signal concerning the vibration so that the non-linear distortion which occurs from the loudspeaker is removed and so that a frequency characteristic concerning the vibration of the loudspeaker becomes a predetermined frequency characteristic.

Abstract: A signal-processing circuit for the generation of a loudspeaker signal comprises an input for the feeding of a digital input signal and a digital equalizer filter that is coupled with the input and has at least one first recursive filter that is defined by a first adjustable set of coefficients. An amplification device coupled with the equalizer filter is designed to generate the loudspeaker signal as a function of the filtered input signal and to output on an output terminal. A current measurement device is designed to output a digital measurement signal that represents a current of the loudspeaker signal. A digital filter block is designed to generate, through filtering, an estimate signal as a function of the measurement signal. For this purpose, the filter block has at least one second recursive filter that is defined by a second adjustable set of coefficients.

Abstract: A signal processing circuit includes a delaying unit that is configured to carry out delay processing on the basis of periodicity information synchronized with the periodicity of a periodic noise included in an input signal and a filter unit that is configured to receive the input signal and has a notch characteristic at a frequency f. The frequency f satisfies f=N/T, where N represents an integer equal to or greater than one and T represents a delaying time applied by the delaying unit.

Abstract: Noise effects in a signal for driving a plant are reduced by generating a reference signal from the error signal. A signal generator generates a reference signal for input to a finite impulse response (FIR) filter. The error signal is produced by differencing the transfer function output and a disturbance signal. The error signal is input to the signal generator and to a least mean square calculator. The reference signal is input to a copy of the transfer function that outputs a modified reference signal. The modified reference signal is input to least mean square calculator. An LMS signal that updates the filter coefficients to minimize the mean square error is calculated and the LMS signal and the reference signal are input to the FIR filter with the FIR filter being arranged to process the LMS signal and the reference signal to minimize the error signal.

Abstract: A method of analyzing time coherence in the noisy signal including the steps of: a) determining a reference signal from the noisy signal by applying treatment (10, 18) to the noisy signal that is suitable for attenuating speech components more strongly than the noise component, in particular by an adaptive recursive predictive algorithm of the LMS type; b) determining (24) a probability of speech being present/absent on the basis of the respective energy levels in the spectral domain of the noisy signal and of the reference signal; and c) deriving (26) a denoised estimate of the speech signal from the noise signal as a function of the probability of the speech being present/absent as determined in this way.

Abstract: Active noise control system and method for controlling an acoustic noise generated by a noise source at a listening location, in which system and method sound is picked up in the surroundings of the listening location by a sound sensor; an electrical noise signal which corresponds to the acoustic noise of the noise source is generated and filtered adaptively in accordance with control signals. The adaptively filtered noise signal is irradiated into the surroundings of the listening location by a sound reproduction device, where a secondary path transfer function extends between the sound reproduction device and sound sensor. The noise signal is filtered with a transfer function that models the secondary path transfer function. The signals which are provided by the sound sensor after first filtering serve as control signals for the adaptive filtering.

Abstract: Disclosed are a method and a device for suppressing vibrations (18) in an installation comprising an actuator (14) for actuating a flap (12) or a valve (70) used for metering a gas or liquid volume flow (16), especially in the area of HVAC, fire protection, or smoke protection. Vibrations (18) of the flap (12) or valve (70) caused by an unfavorable or wrong adjustment or configuration of the controller and/or by disruptive influences are detected and dampened or suppressed by means of an algorithm (1) that is stored in a microprocessor (49). Said algorithm is preferably based on the components recognition of vibrations (46), adaptive filtering (48), and recognition of sudden load variations (50).

Abstract: Method for processing speech signal data. A speech signal is divided into frames. Each frame is characterized by a frame number T representing a unique interval of time. Each speech signal is characterized by a power spectrum with respect to frame T and frequency band ?. A speech segment and a reverberation segment of the speech signal is determined. L filter coefficients W(k) (k=1, 2, . . . , L) respectively corresponding to L frames immediately preceding frame T are computed such that the L filter coefficients minimize a function ? that is a linear combination of sum of squares of a residual speech power in the reverberation segment and a sum of squares of a subtracted speech power in the speech segment. The computed L filter coefficients are stored within storage media of the computing apparatus.

Abstract: An Active Noise Control (ANC) for controlling a noise produced by a noise source may include an acoustic sensor (212) to sense a noise pattern and to produce a noise signal corresponding to the sensed noise pattern, an estimator (202) to produce a predicted noise signal by applying an estimation function to the noise signal, and an acoustic transducer (216) to produce a noise destructive pattern based on the predicted noise signal.

Abstract: A method for varying speech speed is provided. The method includes the following steps: receive an original speech signal; calculate a pitch period of the original speech signal; define search ranges according to the pitch period; find a maximum within each of the search ranges of the original speech signal; divide the original speech signal into speech sections according to the maxima; obtain a speed-varied speech signal by applying a speed-varying algorithm to each speech section of the original speed signal according to a speed-varying command; and eventually, output the speed-varied speech signal.

Abstract: There is provided a noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

Abstract: An adaptive feedback estimation and cancellation (AFEC) apparatus includes: a controller for generating and outputting control information by using a synchronization signal from an external synchronization acquisition unit and base station information, in order to remove a feedback signal that exists in a forward/reverse repeater signal to be repeated and then send the forward/reverse repeater signal; a first feedback prediction canceller for adaptively removing a feedback signal that exists in the forward repeater signal based on the control information from the controller and automatically adjusting the gain of the forward repeater signal; and a second feedback prediction canceller for adaptively removing a feedback signal that exists in the reverse repeater signal based on the control information from the controller and automatically controlling the gain of the reverse repeater 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: A: for each speaker hpj, at least one sound signal S is sent on the speaker hpj, for each microphone mi, a piece of information hpjmi is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hpj and the microphone mi, B: a reference matrix Qr is saved, this reference matrix being constituted by all the pieces of reference information hpjmi obtained following the sending of the sound signal S, C: as soon as a comparison is to be made, the step A is run with a sound signal S? to obtain current information on a matrix Q, D: the matrices Q and Qr are compared.

Abstract: Methods, systems, and apparatus, including computer program products, for restoring audio signals. A data sequence of samples representing an audio signal is received. Multiple filter coefficients are defined for a filter, and a current sample in the data sequence is selected to be processed. The filter coefficients are updated based on a previous sample preceding the current sample in the data sequence and a filtered value determined by the filter for the previous sample. A filtered value for the current sample is determined using the filter with the updated filter coefficients. The filtered value of the current sample is used to determine whether the current sample has been corrupted by impulsive noise, for example, a crackle.

Abstract: A signal processor includes: a first adaptive filter that takes a first signal as input and generates a first pseudo signal; a first subtractor that subtracts the first pseudo signal from a second signal to supply a first differential signal as output; a second adaptive filter that takes the first signal as input to generate a second pseudo signal; a second subtractor that subtracts the second pseudo signal from the second signal to supply a second differential signal as output; a first step size control circuit that generates a first step size used in updating the first adaptive filter in accordance with the relation between the second pseudo signal and the second differential signal; and a second step size control circuit that generates a second step size used in updating the second adaptive filter in accordance with the relation between the first signal and the second signal.

Abstract: One embodiment of the present invention provides a system that cancels fan noise in a computer system. During operation, the system obtains a fan noise signal using a microphone. Next, the system generates a spectral pattern based on the obtained fan noise signal. The system then uses the spectral pattern to identify a corresponding cancellation spectrum in an anti-spectra library. Next, the system generates a noise-canceling signal using the cancellation spectrum. Note that the amount of computation required to cancel fan noise is reduced because generating the noise-canceling signal using the anti-spectra library requires less computation than generating the noise-canceling signal using dynamic noise-cancellation techniques.

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: An active noise control apparatus for reducing noise from a noise source includes a microphone for detecting noise produced by the noise source, and a generalized finite impulse response (FIR) filter for receiving noise signals of the detected noise from the microphone and generating control signals for reducing the noise from the noise source. A speaker produces sound based on the control signals from the generalized FIR filter for substantially canceling the noise from the noise source.