Abstract: An audio apparatus has a function of correcting an audio signal in response to a noise level. The audio apparatus includes a correction unit that corrects an input audio signal on the basis of a weighting factor, an output unit that produces a played-back audio sound on the basis of the corrected audio signal, a microphone for receiving an external sound that includes the played-back audio sound and noise, a noise-extracting unit that extracts a noise signal from an external sound signal, the noise-extracting unit including a speech-removing unit that removes a speech signal from the noise signal on the basis of noise spectrum data, and a weighting factor calculation unit that calculates the weighting factor on the basis of the extracted noise signal and supplies the calculated weighting factor to the correction unit.

Abstract: In signal-component extraction, an input signal is delayed to generate a delayed input signal. The input signal is adaptively filtered with filter coefficients, to generate a filtered signal. The filtered signal is subtracted from the delayed input signal to generate an error signal. A preset reference value is divided by an amplitude of the input signal to generate a gain value. The filter coefficients are derived based on a value obtained by multiplying the input signal and error signal by the gain value or a square of the gain value.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: A method and system to reduce the noise floor of a communications system is disclosed. The system may be incorporated into any device that provides binary samples from a datastream, such as a cordless telephone system. The system is configured to determine a number of bits of the binary samples that are affected by noise. The system is then able to remove the noise by setting those bits to a fixed value. The fixed value may depend on whether the sample is positive or negative. The value to set may be chosen so that the least significant bits of each sample come as close as possible to 0 for that particular numerical representation system. The system can be integrated with other known signal processing methods.

Abstract: Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

Abstract: Disclosed herein is a signal processing apparatus including: a first decimation processing section for generating, based on a digital signal in a first form, a digital signal in a second form; a second decimation processing section for generating, based on the digital signal in the second form, a digital signal in a third form; a first signal processing section for processing the digital signal in the third form; an interpolation processing section for converting a digital signal in the third form outputted from the first signal processing section into a digital signal in the second form; a second signal processing section for processing the digital signal in the second form outputted from the first decimation processing section; and a combining section for combining the digital signals in the second form outputted from the interpolation processing section and the second signal processing section.

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: Masking thresholds are obtained for each frequency component of sound data and ambient noise. It is determined whether each frequency component of the sound data is masked by at least one of the other frequency components of the sound data. It is further determined whether each frequency component of the sound data is masked by ambient noise. Correction coefficients are set for each frequency component of the sound data according to whether the frequency component is masked by at least one of the other frequency components of the sound data and whether the frequency component is masked by the ambient noise. And each frequency component of the sound data is corrected by using the respective correction coefficients.

Abstract: A signal separator, a method and computer product for determining a first output signal describing an audio content of a useful-signal source in a first microphone signal, and for determining a second output signal describing an audio content of the useful-signal source in a second microphone signal.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: A method and apparatus for detecting microphone interference includes first and second built-in microphones producing first and second microphone signals. A first filter bank creates first high-frequency-band and first low-frequency-band signals from the first microphone signal. A second filter bank creates second high-frequency-band and second low-frequency-band signals from the second microphone signal. A first measurement calculator determines a high-frequency-band energy value from the first high-frequency-band signal and the second high-frequency-band signal when the first and second high-frequency-band signals' magnitudes exceeds predetermined thresholds. A second measurement calculator calculates a low-frequency-band energy value from the first low-frequency-band signal and the second low-frequency-band signal when the first and second low-frequency-band signals' magnitudes exceed predetermined thresholds.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.

Abstract: A dereverberation device includes a reverberation estimation unit for estimating a later reflection component by using information on an impulse response from a signal source to an observation point, a noise estimation unit, and a mixing unit. As a result, it is possible to obtain a high-quality dereverberated signal with a small calculation amount even in a noisy environment.

Abstract: Disclosed is an apparatus and method for processing signals such as sound signals. The sound processing apparatus includes a sound signal input unit for receiving sound signals, a harmonic noise separator for separating a harmonic region and a noise region from the received sound signals, a noise restraint index determination unit for determining an optimal noise restraint index k according to a system and circumstance, and a noise restrainer for restraining the separated noise region depending on the noise restraint index k so as to output noise attenuated signals.

Abstract: A hands-free unit comprises a noise tolerant audio sensor to generate a first audio signal based on detection of audible sounds and an external audio sensor to generate a second audio signal based on detection of the audible sounds. A tunable distortion reduction filter adds high frequency information to the first audio signal and reduces distortion. A control unit detects noise levels based on comparison of first and second audio signals; and selects one of the first and second audio signals based on the detected noise level.

Abstract: A transient signal encoding method and device, decoding method and device, and processing system, where the transient signal encoding method includes: obtaining a reference sub-frame where a maximal time envelope having a maximal amplitude value is located from time envelopes of all sub-frames of an input transient signal; adjusting an amplitude value of the time envelope of each sub-frame before the reference sub-frame in such a way that a first difference is greater than a preset first threshold, in which the first difference is a difference between the amplitude value of the time envelope of each sub-frame before the reference sub-frame and the amplitude value of the maximal time envelope; and writing the adjusted time envelope into bitstream.

Abstract: Apparatus for generating a first acoustic signal and simultaneously sensing a second acoustic signal. The apparatus comprises: an input (15) for receiving a first electrical signal from a signal source; a loudspeaker terminal, directly or indirectly connected to the input (15), for connection to a loudspeaker (30) for generating the first acoustic signal in response to the first electrical signal and for generating a second electrical signal in response to the second acoustic signal; and an output (35), for outputting the second electrical signal. The loudspeaker terminal is connected to the output (35) via isolation means (25) comprising a first filter which is adapted to suppress a signal component related to the first electrical signal while simultaneously passing the second electrical signal to the output (35).

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: The present application relates to a hearing aid with suppression of wind noise wherein wind noise detection is provided involving only a single comparison of the input signal power level at first low frequencies with the input signal power level at frequencies that may include the first low frequencies whereby a computational cost effective and simple wind noise detection is provided. The determination of relative power levels of the input signal reflects the shape of the power spectrum of the signal, and the detection scheme is therefore typically capable of distinguishing music from wind noise so that attenuation of desired music is substantially avoided.

Abstract: A signal processing apparatus includes a signal transforming unit transforming a first sound signal obtained by dividing a sound signal represented by a time function and by the predetermined time width into a second sound signal represented by a frequency function; a calculating unit determining a third sound signal which has a reduced influence of the operation sound by using the second sound signal and a sound signal representing the operation sound; a correcting unit performing a correction, by setting a sound signal representing the target sound as a reference signal, on the third sound signal based on the reference signal; and a signal inverse transforming unit inverse-transforming a sound signal on which the correction is performed from the sound signal represented by the frequency function into the sound signal represented by the time function.

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

Abstract: 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 method for processing an audio signal received through a microphone array coupled to an interfacing device is provided. The method is processing at least in part by a computing device that communicates with the interfacing device. The method includes receiving a signal at the microphone array and applying adaptive beam-forming to the signal to yield an enhanced source component of the signal. Also, an inverse beam-forming is applied to the signal to yield an enhanced noise component of the signal. The method combines the enhanced source component and the enhanced noise component to produce a noise reduced signal, where the noise reduced signal is a target voice signal. Then, monitoring an acoustic set-up associated with the audio signal as a background process using the adaptive beam-forming inverse beam-forming to track the target signal component, and periodically setting a calibration of the monitored acoustic set-up.

Abstract: The invention relates to a method for assessing interfering noise in motor vehicles, according to which noise occurring during a predefined measuring time is divided into different frequency ranges, the changes in level relative to the background noise are determined within said frequency ranges, and the determined changes in level are evaluated.

Abstract: Systems and methods provided herein decrease an idle channel noise floor and reduce power during an idle channel input for low power audio devices that include a digital pulse width modulation (PWM) amplifier having a noise shaper. An audio data signal is monitored for an idle channel condition. The noise shaper performs quantization of the audio data signal and uses noise shaper filter coefficients to shape noise resulting from the quantization. Predetermined values for the noise shaper filter coefficients are used to shape the noise resulting from quantization while the idle channel condition is not being detected. The values of the noise shaper filter coefficients are reduced so that the values move toward zeros, and the reduced values of the noise shaper filter coefficients are used to attenuate noise resulting from quantization, while the idle channel condition is being detected.

Abstract: A mixed audio separation system (100) which separates a specific audio from among a mixed audio (S100) includes a local frequency information generation unit (105) which obtains pieces of local frequency information (S103) corresponding to local reference waveforms (S102), based on the local reference waveforms (S102) and an analysis waveform which is the waveform of the mixed audio (S100). Each of the local reference waveforms (S102) (i) constitutes a part of a reference waveform for analyzing a predetermined frequency, (ii) has a predetermined temporal/spatial resolution and (iii) includes at least one of an amplification spectrum and a phase spectrum in the predetermined frequency.

Abstract: The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u 1, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (f1(-t)) a first one of the input audio signals (u1) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (f1(-t)) and the second filter (f2(-t)) being adaptable with a first step size (a1) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (u1, u2) a first noise measure (x1) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (a1, (x2) with an equation comprising in a denominator the first noise measure (x1) for the first step size (a1), respectively the

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: 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 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: 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: 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: 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 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: 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: 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: 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.