Abstract: In a listening device such as for example a hearing aid (1) where an input signal (10) is received by a microphone (2), converted from analog to digital (3), digitally processed (4) including a conversion from a time domain into a frequency domain, converted from digital to analog (5) and transmitted to a user by means of a loudspeaker (6), the internal digital processing (4) generates an unwanted noise signal, the so called undesired periodic noise (12), at specific frequencies. The undesired periodic noise is coupled via ground and the battery (7) into the signal processing path. According to the invention, the undesired periodic noise is filtered out of the input signal (10.2) during the digital signal processing (4), after the conversion of the digital signal into the frequency domain.

Abstract: A capacitive micro-electromechanical system (MEMS) microphone includes a semiconductor substrate having an opening that extends through the substrate. The microphone has a membrane that extends across the opening and a back-plate that extends across the opening. The membrane is configured to generate a signal in response to sound. The back-plate is separated from the membrane by an insulator and the back-plate exhibits a spring constant. The microphone further includes a back-chamber that encloses the opening to form a pressure chamber with the membrane, and a tuning structure configured to set a resonance frequency of the back-plate to a value that is substantially the same as a value of a resonance frequency of the membrane.

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: A noise suppressor selects, for individual frequency components, maximums by comparing a plurality of noise suppressed spectra 105 and 106 a plurality of noise suppressing units 4 and 5 output, thereby obtaining an output spectrum 107 having the frequency components selected as its components. A first noise suppressing unit 4 generates a noise suppressed spectrum 105 by multiplying an input spectrum 102 by amplitude suppression gains, and makes the amplitude suppression gains greater than most of the amplitude suppression gains in a noise signal intervals of a second noise suppressing unit 5.

Abstract: The invention provides an Internet communication device. The Internet communication device plays a remote audio signal received via a network and transmits an audio signal back to the remote party to complete the communication. The Internet communication device comprises a line-in speech detection module and a line-in channel control module. The line-in speech detection module detects whether the remote audio signal is speech or not to generate a remote speech detection result. The line-in channel control module then attenuates the remote audio signal if the remote speech detection result indicates that the remote audio signal is not speech, thus, all noise including non-stationary noise is removed from the remote audio signal.

Abstract: There are included an LPF (3) and an HPF (4) that are connected in parallel to the output of a pre-emphasis circuit (2). There is also included a gain adjusting circuit (6) that performs a gain adjustment of low-pass filter with respect to the frequency band to be passed through the HPF (4). The low frequency components of the frequency band of baseband signals outputted from the pre-emphasis circuit (2) pass through the LPF (3), while the high frequency components pass through the HPF (4). As to the outputs from the HPF (4), the gain of especially the higher part of the frequency band components to be passed through the HPF (4) is suppressed by the gain adjusting circuit (6), whereby the amplitudes of the baseband signals can be limited only for the high frequency range without using a limiter and further the peak values of the baseband signals can be inhibited from exceeding the maximum frequency deviation.

Abstract: A noise reduction system is used in a BTSC system to reduce noise of an audio signal. The noise reduction system has an audio spectral compressing unit that has a filter and a memory in the approach of the digital processing. The filter is arranged to filter an input signal according to a transfer function, a variable d, and several parameters b0/a0, a0/b0, b1/b0 and a1/a0. The memory is arranged to store the parameters.

Abstract: A transient in a digital audio signal can be detected by generating a first set of spectral characteristics associated with a first portion of the digital audio signal and a second set of spectral characteristics associated with a second portion of the digital audio signal, wherein the first and second portions of the digital audio signal partially overlap, comparing values in the first set of spectral characteristics with corresponding values in the second set of spectral characteristics to generate a set of ratios, weighting the set of ratios, and analyzing at least a portion of the weighted set of ratios to detect a transient associated with the first portion of the digital audio signal. Further, an indicator identifying the presence of a detected transient can be output. Additionally, one or more ratios in the set of ratios can be weighted based on amplitude, frequency, or a power function.

Abstract: Various embodiments of systems and methods for reducing audio noise are disclosed. One or more sound components such as noise and network tone can be detected based on power spectrum obtained from a time-domain signal. Results of such detection can be used to make decisions in determination of an adjustment spectrum that can be applied to the power spectrum. The adjusted spectrum can be transformed back into a time-domain signal that substantially removes undesirable noise(s) and/or accounts for known sound components such as the network tone.

Abstract: An exemplary method of reducing an effect of ambient noise within an auditory prosthesis system includes dividing an audio signal presented to an auditory prosthesis patient into a plurality of analysis channels each containing a frequency domain signal representative of a distinct frequency portion of the audio signal, determining a signal-to-noise ratio and a noise reduction gain parameter based on the signal-to-noise ratio for each of the frequency domain signals, applying noise reduction to the frequency domain signals in accordance with the determined noise reduction gain parameters to generate a noise reduced frequency domain signal corresponding to each of the analysis channels, and generating one or more stimulation parameters based on the noise reduced frequency domain signals and in accordance with at least one of a current steering stimulation strategy and an N-of-M stimulation strategy. Corresponding methods and systems are also disclosed.

Abstract: A method of interference suppression is provided that includes receiving a first audio signal from a first audio capture device and a second audio signal from a second audio capture device wherein the first audio signal includes a first combination of desired audio content and interference and the second audio signal includes a second combination of the desired audio content and the interference, performing blind source separation using the first audio signal and the second audio signal to generate an output interference signal and an output audio signal including the desired audio content with the interference suppressed, estimating interference remaining in the output audio signal using the output interference signal, and subtracting the estimated interference from the output audio signal to generate a final output audio signal with the interference further suppressed.

Abstract: A method of suppressing noise in a signal containing speech and noise to provide a noise suppressed speech signal. An estimate is made of the noise and an estimate is made of speech together with some noise. The level of the noise included in the estimate of the speech together with some noise is variable so as to include a desired amount of noise in the noise-suppressed signal.

Abstract: An approach is provided for measuring, identifying, and removing at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in a noise signal (w(t), w(?·?t)). A frequency range to be measured is split into a plurality of frequency bands (?) via a Fast Fourier Transform (FFT) filter bank. For each of the frequency bands (?), an autocorrelation matrix ({circumflex over (R)}?) is determined, wherein parameters of the autocorrelation matrices ({circumflex over (R)}?) are variably adjusted based on whether the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) is to be measured, identified, or removed and further based on at least one averaging. The autocorrelation matrices ({circumflex over (R)}?) are jointly utilized for one or more of measuring, identifying, or removing the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in the noise signal (w(t), w(?·?t)).

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: A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment.

Abstract: An audio enhancement system is provided for compensating for distortions (e.g., linear distortions) of a sound signal reproduced by an audio system in a listening room. The audio enhancement system includes analysis filters that generate a plurality of analysis output signals from an audio signal to be enhanced. The system also includes synthesis filters that generate an enhanced audio signal from a number of synthesis input signals. The number of analysis output signals and the number of synthesis input signals preferably are equal. Signal processing elements between the analysis filters and the synthesis filters generate one of the synthesis input signals from a respective one of the analysis output signals to perform an inverse filtering for linearizing an unknown transfer function indicative of the audio system and the listening room in the respective frequency range.

Abstract: The present invention discloses a multi-level output signal converter, which is connected to an audio amplifier. The audio amplifier comprises a comparing/measuring device, an encoder and an output unit. The multi-level output signal converter comprises a timing processing unit and a multi-level converter. The timing processing unit is connected to the comparing/measuring device and the encoder. The timing processing unit includes a plurality of flip-flops and a timing summing element. The flip-flop receives a first signal from the comparing/measuring device and outputs the first signal to the timing summing element. The encoder converts the first signal into a second signal. The multi-level converter is connected to the encoder and the output unit. The encoder transmits the second signal to the multi-level converter, and the multi-level converter thus outputs a third signal to the output unit.

Abstract: There is provided a signal processing apparatus, for suppressing a noise, which includes a first calculator to obtain a phase difference between two spectrum signals in a frequency domain transformed from sound signals received by at least two microphones to estimate a sound source by the phase difference, a second calculator to obtain a value representing a target signal likelihood and to determine a sound suppressing phase difference range at each frequency, in which a sound signal is suppressed, on the basis of the target signal likelihood, and a filter. The filter generate a synchronized spectrum signal by synchronizing each frequency component of one of the two spectrum signals to each frequency component of the other of the two spectrum signals for each frequency when the phase difference is within the sound suppressing phase difference range and to generate a filtered spectrum signal.

Abstract: An acoustic system that eliminates the howling that occurs when the sound outputted by the speaker feeds back to the input device. The acoustic system comprises a digital signal processor (DSP) that divides the input audio signal into different frequency bands, and reduces the audio levels for the frequency bands where howling is most likely to occur. In one embodiment, the acoustic system comprises a sound source section that generates a test tone that substantially covers the entire human audible range such that the DSP can set the filter levels according to the feedback of the test tone. In another embodiment, the sound source section stores one waveform at a given pitch and generates waveforms of other pitches based on the stored waveform. In yet another embodiment, the pitches of the generated waveforms are dispersed into four frequency bands to create a test tone that resembles a chord or a musical tone.

Abstract: A method for adjusting the volume and frequency response for the audio output in a mobile communication device comprises estimating the noise level of the environment surrounding the mobile communication device and then adjusting the volume and frequency response based on the estimated noise level.

Abstract: A microphone array system and a method implemented therefore are provided. A first microphone having a first sensibility receives a sound source to generate a first signal. A second microphone is deposited at a distance from the first microphone, having a second sensibility for receiving the sound source to generate a second signal. A comparator subtracts the first signal and the second signal to generate a difference signal. An analyzer estimates an incident angle of the sound source to determine a compensation factor based on the first signal and the difference signal. A gain stage adjusts a gain of the difference signal based on the compensation factor to output an output signal.

Abstract: A method and an apparatus for removing white noise in a portable terminal are provided. The method for removing the white noise in the portable terminal includes measuring a volume variation of a voice signal output from a power amplifier; detecting a frequency band including white noise using the measured volume variation; and removing signals of the detected frequency band in the voice signal before output to speaker.

Abstract: A system identifying device for identifying an unknown system interposed between first and second input terminals.

Abstract: A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well.

Abstract: In a method and system for identifying speech sound and non-speech sound in an environment, a speech signal and other non-speech signals are identified from a mixed sound source having a plurality of channels. The method includes the following steps: (a) using a blind source separation (BSS) unit to separate the mixed sound source into a plurality of sound signals; (b) storing spectrum of each of the sound signals; (c) calculating spectrum fluctuation of each of the sound signals in accordance with stored past spectrum information and current spectrum information sent from the blind source separation unit; and (d) identifying one of the sound signals that has a largest spectrum fluctuation as the speech signal.

Abstract: A noise attenuation system attenuates noise in an input signal. The system may estimate a power of the input signal, and determine a noise power value based on the input power estimate. The noise power value corresponds to an estimate of a noise power within the input signal. The system may determine an attenuation factor based on the noise power value, and attenuate the input signal by using the attenuation factor.

Abstract: A novel system prevents surrounding sound to enter through a hearing apparatus, for instance through a ventilation opening, and reach an eardrum of the wearer in the form of interference sound. Contrary to auditory accessories designed especially to protect against noise, it is not possible for many hearing apparatus to compensate for such an interference sound by means of active noise cancellation. The hearing apparatuses do not have the special components needed. No compensation sound signal can therefore form with a correct phase. In accordance with the invention, a compensation sound is only generated for a relatively narrow spectral band. This spectral band is determined as a function of a hearing ability of the wearer of the hearing apparatus and/or as a function of a spectral distribution of the energy of the interference sound or a sound producing the interference sound. The improvement is particularly suited to compensating for an interference sound in a hearing device.

Abstract: Provision to reduce production of musical noise. A noise reduction device includes: means for calculating a rank for each element included in a first region having predetermined sizes in the time axis direction and in the frequency axis direction, depending on a value of the element, in a noise section of an observed signal indicating variation of a frequency spectrum with time; means for calculating a rank for each element included in a second region, depending on a value of the element, the second region having predetermined sizes in the time axis direction and in the frequency axis direction in the observed signal; and means for subtracting, from the values of the respective elements in the second region, values based on the values of the respective elements in the first region whose ranks correspond to ranks of respective elements in the second region.

Abstract: A sound recording device includes a disk drive, a microphone, and a controller. The disk drive writes data on a rotatable data storage disk therein. The microphone generates a microphone signal which includes a desired sound component and a noise component from noise that is generated by the disk drive. The controller filters the microphone signal to reduce the noise component from the disk drive relative to the desired sound component, and writes the filtered microphone signal on the disk. Accordingly, noise from the disk drive can be attenuated during recording by the sound recording device.

Abstract: A noise suppressing device includes a plurality of sound input units inputting sounds from a given sound source and converting the sounds to sound signals on a time axis, a transfer characteristic obtaining unit performing frequency transform of the sound signals after dividing the sound signals into frames and calculating respective transfer characteristics of the sounds for each given frequency band, a storage unit storing the calculated transfer characteristics of the sounds, a frequency obtaining unit obtaining a frequency for updating the transfer characteristics stored in the storage unit for the frequency band, an updating unit updating the transfer characteristics every given number of frames corresponding to the obtained frequency based on the transfer characteristics for each frequency band, a generating unit generating suppression information for suppressing the noise component based on the updated transfer characteristics, and a suppression unit suppressing the noise component based on the suppres

Abstract: A noise removal device includes: an FFT analysis unit which receives a mixed sound including to-be-extracted sounds and noises, and determines frequency signals at time points in a time width; and a to-be-extracted sound determination unit which determines, for each to-be-extracted sound, frequency signals at the time points, satisfying conditions of (i) being equal to or greater than a first threshold value in number and (ii) having a phase distance between the frequency signals that is equal to or smaller than a second threshold value, wherein the phase distance is a distance between phases ??(t) of the condition-satisfying frequency signals when a phase of a frequency signal at a current time point t is ?(t) (radian) and the phase ??(t) is mod 2?(?(t)?2?ft), f denoting a reference frequency, and the predetermined time width is within 2 to 4 times the time window widths of the window functions.

Abstract: A method and system for automatic gain control of a speech signal in a communication system are disclosed. The gain of the speech signal can be controlled, based on a calculated gain value. This gain value is calculated on the basis of energy calculation and speech activity identification in the speech signal which is done by means of the encoder. Encoding the gain controlled speech signal for transmission follows the step of gain control.

Abstract: There is provided a signal processing device including: an audio signal acquisition portion that acquires audio signals; an external signal acquisition portion that acquires external signals; an output signal generation portion that generates output signals from the audio signals and the external signals; a mode setting portion that sets an external mode as an operation mode; and a fade control portion that controls the output signal generation portion in accordance with the operation mode. When the external mode is set, the fade control portion causes the output signal generation portion to generate the output signal for one of the right ear and the left ear of the user from at least the external signal, and also to generate the output signal for the other ear from at least the audio signal.

Abstract: Disclosed is a channel estimator and a method for changing a coefficient of an IIR filter depending on a moving speed of a mobile communication terminal. In the channel estimator, a coefficient changing unit receives I and Q signals from a current base station, and selects a coefficient of the IIR filter optimized depending on the moving speed of the current mobile communication terminal. The coefficient changing unit sets the selected coefficient of the IIR filter to the IIR filter of the channel estimator. Accordingly, it is possible to prevent the performance degradation of the channel estimator caused by the speed of the mobile communication terminal.

Abstract: Techniques pertaining to techniques to reduce wind noises effectively in recorded signals are disclosed. According to one aspect of the present invention, there is a strong correlation between two voice signals from target voices in the same frequency band sampled simultaneously by a pair of microphones in a common scene while there is a weak correlation between wind noises in the same frequency band of the two voice signals sampled simultaneously by the pair of microphones in the common scene. Taking advantage of this feature to provide a larger gain to the frequency band having a strong correlation and a smaller gain to the frequency band having a weak correlation, thereby the wind noise is reduced efficiently with minimum impact on the target voices.

Abstract: A method and apparatus for detecting a singing frequency in a signal processing system using two neural-networks is disclosed. The first one (a hit neural network) monitors the maximum spectral peak FFT bin as it changes with time. The second one (change neural network) monitors the monotonic increasing behavior. The inputs to the neural-networks are the maximum spectral magnitude bin and its rate of change in time. The output is an indication whether howling is likely to occur and the corresponding singing frequency. Once the singing frequency is identified, it can be suppressed using any one of many available techniques such as notch filters. Several improvements of the base method or apparatus are also disclosed, where additional neural networks are used to detect more than one singing frequency.

Abstract: A method and apparatus identify a clean speech signal from a noisy speech signal. To do this, a clean speech value and a noise value are estimated from the noisy speech signal. The clean speech value and the noise value are then used to define a gain on a filter. The noisy speech signal is applied to the filter to produce the clean speech signal. Under some embodiments, the noise value and the clean speech value are used in both the numerator and the denominator of the filter gain, with the numerator being guaranteed to be positive.

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: A method of denoise a stereo signal comprising a stereo sum signal and a stereo difference signal, performs a frequency selective stereo to mono blending based on the masking effect of the human auditory system. Therefore, a stereo signal noise reducer, comprising a first filter bank (1) to split the stereo difference signal (l?r) into a plurality of subbands, respective first multipliers (CO, . . . , CN) to weight each of the subbands of the stereo difference signal with a respective corresponding control signal (CO, . . . , CN), and a first adder (3) to sum all weighted subbands of the stereo difference signal (l?r) to build a frequency selective weighted stereo difference signal (diff), within which a number and width of the subbands obtained via the first filter bank (1) are choosen according to the properties of the human auditory system, further comprises a weighting factor determination unit which determines a respective control signal (CO, . . .

Abstract: In a noise suppression device, an audio detector detects presence or absence of audio in an input signal. A first noise spectrum estimator estimates a noise spectrum contained in the input signal based on the input signal and detection result of the audio detector. A second noise spectrum estimator estimates the noise spectrum based on the input signal regardless of the detection result of the audio detector. A noise spectrum calculator calculates a final noise spectrum estimation value according to a length of detecting time during which the audio detector continuously detects the audio and based on first and second noise spectrum estimation values that are obtained as estimation results by the first and second noise spectrum estimators. A gain calculator calculates a noise suppression gain based on the final noise spectrum estimation value. A noise suppressor suppresses noise contained in the input signal by applying the noise suppression gain to the input signal.

Abstract: Provided are an apparatus and method for estimating noise and a noise reduction apparatus employing the same. The noise estimation apparatus estimates noise by blocking audio signals from a direction of a target sound source from received audio signals, and compensating for distortions from directivity gains of a target sound blocker blocking the audio signals from the target sound source.

Abstract: Provided is a sound processing device including: a sound input unit for dividing an input sound into predetermined time units; a sound processing unit for encoding the input sound thus divided; a noise detecting unit; and an output control unit for replacing encoded data on the input sound with silent data according to detection results of the noise detecting unit. Also provided is an input sound processing method including: encoding an input sound; judging whether or not the input sound contains a noise; and replacing a noise portion contained in the encoded input sound with silent data.

Abstract: A method and apparatus for filtering an electromyogram (EMG) signal from a raw signal which includes a contribution from an electrocardiogram (EKG) signal is disclosed. The method includes the steps of estimating an attribute (such as a Fourier transform) of both the EMG contribution to the raw signal and the EKG contribution to the raw signal and, dependent on both frequency spectrums, determining an EMG window in a frequency range and obtaining the EMG signal by passing it through a filter defined by the frequency range. The method is particularly used when monitoring a multi-channel electrical recording from a plurality of electrodes attached to a patient's diaphragm.

Abstract: A method of removing noise and interference from a signal by receiving the signal, calculating a joint time-frequency domain of the signal, estimating instantaneous frequencies of the joint time-frequency domain, modifying each estimated instantaneous frequency, if necessary, to correspond to a frequency of the joint time-frequency domain to which it most closely compares, redistributing the elements within the joint time-frequency domain according to the estimated instantaneous frequencies as modified, computing a magnitude for each element in the joint time-frequency domain as redistributed, plotting the results as the time-frequency representation of the signal, identifying in the plot any noise and interference components in the received signal, eliminating from the redistributed joint time-frequency domain elements that correspond to noise and interference, and recovering a signal devoid of noise and interference from the modified redistributed joint time-frequency domain.

Abstract: A method and system processes a speech signal. A fast Fourier transform is performed on a speech signal to produce a speech signal having a plurality of frequency bands in a frequency domain.

Abstract: An audio signal processing apparatus includes a splitting unit for splitting an audio signal of a first system and another audio signal of a second system into pluralities of frequency band components, a level comparing unit for calculating a level ratio or a level difference between each of the frequency bands of the first system and each of the frequency bands of the second systems, and an output control unit for removing frequency band components whose level ratio or level difference calculated by the level comparing unit is equal and substantially equal to a predetermined value from at least one of the first and second systems.

Abstract: The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.

Abstract: A noise suppression estimation device calculates a noise index value which varies according to kurtosis of a frequence distribution of magnitude of a sound signal before or after suppression of the noise component, the noise index value indicating a degree of occurrence of musical noise after suppression of the noise component in a frequency domain. For example, the noise suppression estimation device calculates first kurtosis of a frequence distribution of magnitude of the sound signal before suppression of the noise component, calculates second kurtosis of a frequence distribution of magnitude of the sound signal after suppression of the noise component, and calculates the noise index value from the first kurtosis and the second kurtosis.

Abstract: It is possible to provide a noise suppression method, device, and program capable of realizing a sound image positioning of an output side corresponding to an input side with a small calculation amount. The device includes a common suppression coefficient calculation unit for receiving conversion outputs from a plurality of channels and calculating a suppression coefficient common to the channels.

Abstract: A method for reducing noise associated with an audio signal received through a microphone sensor array is provided. The method initiates with enhancing a target signal component of the audio signal through a first filter. Simultaneously, the target signal component is blocked by a second filter. Then, the output of the first filter and the output of the second filter are combined in a manner to reduce noise without distorting the target signal. Next, an acoustic set-up associated with the audio signal is periodically monitored. Then, a value of the first filter and a value of the second filter are both calibrated based upon the acoustic set-up. A system capable of isolating a target audio signal from multiple noise sources, a video game controller, and an integrated circuit configured to isolate a target audio signal are included.