Abstract: A special purpose machine measures and modulates communication signals that are parsed into frames. Frames of signals modulated and measured to have certain qualities are deemed to be the result of wind noise. Frames of wind noise are cancelled from further use within a communication system.

Abstract: An audio device comprises a processor, an audio processing module, a high frequency noise generating circuit, a first switch control circuit, a second switch control circuit, a low pass filter circuit and an adder circuit. The audio device generates a watermark data, an original audio signal and a high frequency noise signal. The high frequency noise signal pass through a first switch control circuit and a second switch control circuit according to watermark data. A low pass filter circuit filters the high frequency noise signal received from the first switch control circuit to form a first add data. An adder circuit receives the first add data from the low pass filter circuit, receives the high frequency noise signal from the second switch control circuit as a second add data, and adds the first add data and the second add data to the original audio signal.

Abstract: A noise suppression system reduces low-frequency noise in a speech signal using linear predictive coefficients in an adaptive filter. A digital filter may update or adapt a limited set of linear predictive coefficients on a sample-by-sample basis. The linear predictive coefficients may be used to provide an error signal based on a difference between the speech signal and a delayed speech signal. The error signal represents an enhanced speech signal having attenuated and normalized low-frequency noise components.

Abstract: Online source separation may include receiving a sound mixture that includes first audio data from a first source and second audio data from a second source. Online source separation may further include receiving pre-computed reference data corresponding to the first source. Online source separation may also include performing online separation of the second audio data from the first audio data based on the pre-computed reference data.

Abstract: In a system for estimating the power spectral density of acoustical background noise when the level of a smoothed power spectral density signal increases, an increment value is increased, starting from a minimum increment value, by a predetermined amount until a maximum increment value is reached if at the same time the value of the power spectral density currently determined in a new calculation cycle is larger than the estimate value of the power spectral density of the background noise determined in the previous calculation cycle. For cases in which the level of the smoothed power spectral density decreases, the amplitude of the decrement value is increased, starting from a minimum decrement value, by a predetermined amount until a maximum decrement value is reached if at the same time the value of the power spectral density currently determined in a new calculation cycle is smaller than the estimate value of the power spectral density of the background noise determined in the previous calculation cycle.

Abstract: A method for processing signals for reducing noise components in a vehicular microphone system (500A) compares (517) a plurality of noise component values (511, 513, 515) for at least one frequency band of plurality of frequency bands. At least one frequency band is then downwardly expanded (521) by a predetermined expansion ratio (523) for providing a noise reduced signal (535) when determining a value is below a predetermined threshold (519).

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: Described is a audio signal processing technology in which an adaptive beamformer processes input signals from microphones based on an estimate received from a pre-filter. The adaptive beamformer may compute its parameters (e.g., weights) for each frame based on the estimate, via a magnitude-domain objective function or log-magnitude-domain objective function. The pre-filter may include a time invariant beamformer and/or a non-linear spatial filter, and/or may include a spectral filter. The computed parameters may be adjusted based on a constraint, which may be selectively applied only at desired times.

Abstract: Suppression of interference across transceivers integrated on a single semiconductor chip. An example of a method of reducing noise in a transceiver includes introducing an adjustable time delay into a signal between a first section of a signal path into which noise may be introduced and a second section of the signal path into which noise may be introduced. The method also includes selectively adjusting the time delay and signal polarity to improve a signal-to-noise metric of the transceiver. An example of the transceiver includes a transmitter and a receiver. The transceiver also includes an adjustable time delay between a first section of a transceiver signal path into which noise may be introduced and a second section of the transceiver signal path into which noise may be introduced and circuitry for reducing noise by adjusting a value of the time delay.

Abstract: A method, system, and computer program product for managing noise in a noise reduction system, comprising: receiving a first signal at a first microphone; receiving a second signal at a second microphone; identifying noise estimation in the first signal and the second signal; identifying a transfer function of the noise reduction system using a ratio of a power spectral density of the second signal minus the noise estimation to a power spectral density of the first signal, wherein the noise estimation is removed from only the power spectral density of the second signal; and identifying a gain of the noise reduction system using the transfer function.

Abstract: A pre-coding method for improving diversity gain in an Orthogonal Frequency Division Multiplexing (OFDM) system and a transmission apparatus and method are disclosed. The transmission method includes converting an input transmission signal into parallel signals, pre-coding the parallel signals with a predetermined frequency offset, performing Inverse Discrete Fourier Transform (IDFT) on the pre-coded signals, and up-converting the IDFT signals and outputting the up-converted signals to a wireless network. By performing pre-coding on a transmission signal with a frequency offset in an OFDM system, diversity gain can be improved.

Abstract: A method and device for transforming ambient audio are provided. Example embodiments may include monitoring ambient audio proximate to a sound processing device located in an environment. The device may receive a selection from a user interface. The selection may comprise one of a number of available first selections, each available first selection identifying one of multiple transformation modes. The device may access memory to obtain transformation audio and process the transformation audio, based on the ambient audio and the selection. The device may also use the transformation audio to provide modified output audio for propagation into the environment.

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: The invention provides a method for determining a noise reference signal for noise compensation and/or noise reduction. A first audio signal on a first signal path and a second audio signal on a second signal path are received. The first audio signal is filtered using a first adaptive filter to obtain a first filtered audio signal. The second audio signal is filtered using a second adaptive filter to obtain a second filtered audio signal. The first and the second filtered audio signal are combined to obtain the noise reference signal. The first and the second adaptive filter are adapted such as to minimize a wanted signal component in the noise reference signal.

Abstract: An audio signal processing method for processing input audio signals of plural channels includes calculating at least one feature quantity representing a difference between channels of input audio signals, selecting at least one weighting factor according to the feature quantity from at least one weighting factor dictionary prepared by learning beforehand, and subjecting the input audio signals of plural channels to signal processing including noise suppression and weighting addition using the selected weighting factor to generate output an output audio signal.

Abstract: A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal.

Abstract: A method for separating a sound source from a mixed signal, includes Transforming a mixed signal to channel signals in frequency domain; and grouping several frequency bands for each channel signal to form frequency clusters. Further, the method for separating the sound source from the mixed signal includes separating the frequency clusters by applying a blind source separation to signals in frequency domain for each frequency cluster; and integrating the spectrums of the separated signal to restore the sound source in a time domain wherein each of the separated signals expresses one sound source.

Abstract: In one embodiment, an automated interferometric noise measurement system includes: a signal source adapted to provide a carrier signal; a delay line adapted to delay a first version of the carrier signal to provide a delayed signal to a device-under-test (DUT); a variable attenuator adapted to attenuate a second version of the carrier signal to provide an attenuated signal; a first variable phase-shifter adapted to phase-shift the attenuated signal to provide a first phase-shifted signal; a hybrid coupler adapted to receive an output signal from the DUT and the first phase-shifted signal to provide a carrier-suppressed signal and a carrier-enhanced signal; a low-noise amplifier adapted to amplify the carrier-suppressed signal to provide an amplified signal; a second variable phase-shifter adapted to phase-shift a version of the carrier-enhanced signal to provide a second phase-shifted signal; a first mixer adapted to mix a first version of the amplified signal and the second phase-shifted signal to provide a

Abstract: A method (400, 500) and apparatus (220) seeks to improve the intelligibility of speech emitted into a noisy environment. Formants are identified (426) and perceptual frequency scale band is selected (502) that includes at least one of the identified formants. The SNR in each band is compared (504) to a threshold and, if the SNR for that band is less than the threshold, the method increases a formant enhancement gain for that band. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains yielding combined gains that are then clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532), and used to reconstruct (532, 534) an audio signal.

Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Abstract: The present system proposes a technique called the spectro-temporal varying technique, to compute the suppression gain. This method is motivated by the perceptual properties of human auditory system; specifically, that the human ear has higher frequency resolution in the lower frequencies band and less frequency resolution in the higher frequencies, and also that the important speech information in the high frequencies are consonants which usually have random noise spectral shape. A second property of the human auditory system is that the human ear has lower temporal resolution in the lower frequencies and higher temporal resolution in the higher frequencies. Based on that, the system uses a spectro-temporal varying method which introduces the concept of frequency-smoothing by modifying the estimation of the a posteriori SNR. In addition, the system also makes the a priori SNR time-smoothing factor depend on frequency.

Abstract: A noise reducing apparatus includes: a voice signal inputting unit inputting an input voice signal; a noise occurrence period detecting unit detecting a noise occurrence period; a noise removing unit removing a noise for the noise occurrence period; a generation source signal acquiring unit acquiring a generation source signal with a time duration corresponding to a time duration corresponding to the noise occurrence period; a pitch calculating unit calculating a pitch of an input voice signal interval; an interval signal setting unit setting interval signals divided in each unit period interval; an interpolation signal generating unit generating an interpolation signal with the time duration corresponding to the noise occurrence period and alternately arranging the interval signal in a forward time direction and the interval signal in a backward time direction; and a combining unit combining the interpolation signal and the input voice signal, from which the noise is removed.

Abstract: This specification describes technologies relating to editing audio data. In general, one aspect of the subject matter described in this specification can be embodied in methods that include receiving an audio signal including digital audio data; receiving an input identifying particular audio data of the audio signal corresponding to a noise pulse; and replacing the audio data corresponding to the detected noise pulse using interpolation of adjacent audio data to generate an edited audio signal. Other embodiments of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A noise reducing device includes: a sound-signal input unit that inputs a sound signal; a time-to-frequency converting unit that converts the input sound signal obtained by being input by the sound-signal input unit into a frequency signal; a patterning unit that calculates, for each of divided frequencies of the frequency signal, an nth order polynomial (n is a natural number) as a polynomial for interpolating sampling points and acquires a coefficient pattern including a set of values of coefficients of respective orders of the polynomial; a matching-data storing unit that stores, in association with the divided frequencies, matching data indicating a matching range as a range of a coefficient pattern regarded as noise; and a noise determining unit that determines, on the basis of a result obtained by comparing the coefficient pattern acquired by the patterning unit and the matching range indicated by the matching data, at least presence or absence of noise occurrence at a divided frequency corresponding to

Abstract: In a simulation apparatus, a difference data generation part compares a simulation result of an existing sound generator with a simulation result of a virtual sound generator, and generates difference data representing a difference between the simulation result of the existing sound generator and the simulation result of the virtual sound generator. A characteristic correction part corrects a measurement result of the existing sound generator based on the difference data, and generates virtual sound generator prediction data representing the sound generation characteristic of the virtual sound generator according to the corrected measurement result of the existing sound generator.

Abstract: A noise filling method is provided that includes detecting a frequency band including a part encoded to 0 from a spectrum obtained by decoding a bitstream; generating a noise component for the detected frequency band; and adjusting energy of the frequency band in which the noise component is generated and filled by using energy of the noise component and energy of the frequency band including the part encoded to 0.

Abstract: In a wind noise suppressor, a divider divides the frequency band an input sound into a first frequency band having a possibility that wind noise is included and a second frequency band having a frequency higher than a frequency of the first frequency band, a calculator calculates a probability that the input sound includes wind noise from feature parameters of a sound in the first frequency band, a suppressor suppresses wind noise included in the first frequency band in accordance with an intensity calculated from the probability, and an adder mixes and outputs the sound in the second frequency band divided by the divider and the sound in the first frequency band by which wind noise is suppressed by the suppressor.

Abstract: An audio signal processing apparatus includes: a dividing section dividing each of audio signals of a plurality of channels into a plurality of frequency bands; a phase difference calculating section calculating a phase difference between the audio signals of the plurality of channels, for each of the plurality of frequency bands divided by the dividing section; a level ratio calculating section calculating a level ratio between the audio signals of the plurality of channels, for each of the plurality of frequency bands divided by the dividing section; and an audio signal processing section performing output gain setting with respect to divided signals obtained by the dividing section, on the basis of the phase difference and the level ratio for each of the plurality of frequency bands calculated by the phase difference calculating section and the level ratio calculating section.

Abstract: A noise extraction device of the present invention includes: first and second microphone units (11 and 12) each picking up a sound; a directivity synthesis unit which performs a directivity synthesis on output signals respectively received from the first and second microphone units (11 and 12) and generates two directionally synthesized signals which have: different sensitivities to noise; the same directional pattern with respect to sound pressure; and the same effective acoustic center position; and an acoustic cancellation unit which cancels an acoustic component of one of the two directionally synthesized signals by subtracting the one of the two directionally synthesized signals from the other of the two directionally synthesized signal, so as to extract a noise component.

Abstract: A howling suppression apparatus suppresses a howling caused in an acoustic system including a sound collection device and a sound emission device. An estimation part generates an estimated signal by estimating a feedback sound reaching the sound collection device from the sound emission device. An adjustment part generates an estimated signal by adjusting the estimated signal. A spectrum subtraction part generates an acoustic signal using a result of subtracting a frequency spectrum of the estimated signal from a frequency spectrum of an acoustic signal. A filter part generates an acoustic signal by suppressing a component of a frequency band including a howling frequency F among the acoustic signal. An acoustic signal in which the acoustic signal is amplified by an amplifier is supplied to the sound emission device.

Abstract: A method of enabling a hearer to hear desired sound while also being able to be aware of ambient sound, comprises providing a first non-audio signal representative of said desired sound, deriving a second non-audio signal from said ambient sound, combining the first and second non-audio signals in providing a third non-audio signal, and converting said third non-audio signal into sound.

Abstract: An automatic gain controller and a method using the same are provided. The automatic gain controller and method analyze background noise by operating a microphone mounted in a mobile communication terminal and automatically control the gain of the signal part which is non-audible due to the background noise. Thus, a user may listen to music by using an earphone or a headphone connected to the mobile communication terminal in an environment with background noise. The method includes receiving an audio signal to be reproduced, receiving a background noise signal introduced through a microphone, controlling a gain of the audio signal by comparing the background noise signal and the audio signal and outputting the gain-controlled audio signal so that the user can listen to the gain-controlled audio signal.

Abstract: A method of operating a loudspeaker includes providing a digital audio signal and identifying a target transfer function to be applied to the signal. At least one coefficient of an FIR filter is generated. The generating includes performing Heyser spiral curve fitting, and fitting a three-dimensional curve based on a magnitude and phase of a target transfer function. The digital audio signal is filtered through the FIR filter. The filtered signal is inputted into the loudspeaker.

Abstract: Provided are an apparatus and method for preventing noise. The apparatus estimates a noise signal from a signal transformed into a frequency domain, uses the estimated noise signal to estimate the amplitude of the frequency-domain signal according to a frequency band, and then calculates a phase difference according to a frequency band and eliminates or prevents noise from the amplitude-estimated frequency-domain signal based on the calculated phase difference according frequency band.

Abstract: A noise cancellation apparatus includes a noise estimation module for receiving a noise-containing input speech, and estimating a noise therefrom to output the estimated noise; a first Wiener filter module for receiving the input speech, and applying a first Wiener filter thereto to output a first estimation of clean speech; a database for storing data of a Gaussian mixture model for modeling clean speech; and an MMSE estimation module for receiving the first estimation of clean speech and the data of the Gaussian mixture model to output a second estimation of clean speech. The apparatus further includes a final clean speech estimation module for receiving the second estimation of clean speech from the MMSE estimation module and the estimated noise from the noise estimation module, and obtaining a final Wiener filter gain therefrom to output a final estimation of clean speech by applying the final Wiener filter gain.

Abstract: A signal-processing device includes a determination section that compares a frequency spectrum and a floor spectrum of an input audio signal to each other for each frequency bin and determines whether the input audio signal should be subjected to noise reduction processing or not for each of the frequency bins; and a noise reduction-processing section that subtracts a noise frequency spectrum from the frequency spectrum of the input audio signal for each of the frequency bins on the basis of the result determined by the determination section for each of the frequency bins.

Abstract: A method (400, 600, 700) and apparatus (220) for enhancing the intelligibility of speech emitted into a noisy environment. After filtering (408) ambient noise with a filter (304) that simulates the physical blocking of noise by a at least a part of a voice communication device (102) a frequency dependent SNR of received voice audio relative to ambient noise is computed (424) on a perceptual (e.g. Bark) frequency scale. Formants are identified (426, 600, 700) and the SNR in bands including certain formants are modified (508, 510) with formant enhancement gain factors in order to improve intelligibility. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains factors yielding combined gains which are clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532) and used to reconstruct (532, 534) an audio signal.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving, by an audio processing apparatus, a signal, and feedback information estimated based on a normalizing gain; generating a noise estimation based on the signal; computing a gain filter for noise canceling, based on the noise estimation and the signal; and, obtaining a restricted gain filter by applying the feedback information to the gain filter.

Abstract: According to an aspect of the invention, there is provided a noise suppressing apparatus comprising: a fifth unit configured to calculate a gain for noise suppression, based on the first signal-to-noise ratio for each frequency band and the second signal-to-noise ratio for an entire frequency band; an eighth unit configured to calculate an upper limit value of a noise suppression amount for each frequency band, based on the second signal-to-noise ratio; a ninth unit configured to calculate the noise suppression amount for each frequency band, based on the first signal-to-noise ratio; and a tenth unit configured to limit, based on the upper limit value, the noise suppression amount so as to calculate the gain.

Abstract: Broadly speaking, the embodiments disclosed herein describe an apparatus, system, and method for managing the effects of TDMA noise emitted by a communication device on an audible circuit.

Abstract: A system and method for analyzing a signal to monitor the dynamics of its magnitude and frequency characteristics over time. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention may comprise a feedback control block operable to determine a candidate frequency having potential feedback such that the feedback control block is further operable to perform an iterative analysis of the magnitude of the audio signal at the candidate frequency to determine the growth characteristics of the signal. The electronic circuit may further include a test filter block operable to deploy a test filter at a candidate frequency and a permanent filter block operable to deploy a permanent filter at the candidate frequency if the feedback control block determines that the growth characteristics of the signal at the candidate frequency comprises feedback characteristics after the test filter has been deployed.

Abstract: In one embodiment, one or more users may be participating in a conversation. In one example, a first user may be speaking into a speaker end device and a second user may be listening at a listener end device. The second user may be in an environment where noise may be present. Particular embodiments determine characteristics of the noise at the listener end device. Characteristics of a voice signature for a user speaking with the speaker end device are also determined. Comprehension enhancement of voice signals received from speaker end device is then performed based on characteristics of the noise at the listener end device and characteristics of the voice signature. For example, the signature of the voice signals may be altered to lessen the overlap with the noise.

Abstract: A method for processing an input signal to create an enhanced output signal includes obtaining an envelope of the input signal, determining a logarithm signal of the envelope, determining a rate of change of the logarithm signal to obtain a slope value, and applying a value derived from the slope value to the input signal to thereby generate an enhanced output signal.

Abstract: An audio processing apparatus is provided. A beamformer receives input signals and processes the input signals to generate a first processed signal. The input signals include at least one of a source signal and interference. A blocking matrix receives the input signals and operates to cancel the source signal from the input signals to generate a second processed signal. A first adaptive filter has adaptable first filter coefficients, generates a first filtered signal approximating the interference according to the first and second processed signals and continuously adapts the first filter coefficients according to the first filtered signal and the first processed signal. A second adaptive filter has adaptable second filter coefficients, generates a second filtered signal approximating the interference according to the first and second processed signals and selectively adapts the second filter coefficients according to the first filter coefficients and an output signal.

Abstract: An information processing apparatus includes a feature value detecting section, an image processing section, and an audio processing section. The feature value detecting section determines, when a first image and a second image that are captured at different positions include a specific subject, a feature value of the subject included in the supplied first and second images. The image processing section detects motion of the subject on the basis of the feature value determined by the feature value detecting section. The audio processing section localizes a sound image of the subject in accordance with the motion of the subject detected by the image processing section.

Abstract: According to an embodiment, a method of reducing noise in a signal received at a processing stage of an acoustic system includes, at the processing stage identifying at least one frequency which causes a system gain of the acoustic system to be above an average system gain of the acoustic system; providing a noise attenuation factor for reducing noise in the signal for the at least one frequency, the noise attenuation factor for the at least one frequency based on the system gain for that frequency; and applying the noise attenuation factor to a component of the signal at that frequency.

Abstract: A method of generating parameters representing Head-Related Transfer Functions, the method comprising the steps of a) sampling with a sample length (n) a first time-domain HRTF impulse response signal using a sampling rate (fs) yielding a first time-discrete signal, b) transforming the first time-discrete signal to the frequency domain yielding a first frequency-domain signal, c) splitting the first frequency-domain signal into sub-bands, and d) generating a first parameter of the sub-bands based on a statistical measure of values of the sub-bands.

Abstract: A system and method for analyzing a signal to identify one or more candidate frequencies that may have a potential feedback problem. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention, may comprise a feedback control block operable to: 1) determine a magnitude of a first dominant spectral component in a wide-band span of frequencies of a signal, 2) determine a magnitude of a second dominant spectral component in a mid-band span of frequencies of the signal, and 3) determine a magnitude of a third dominant spectral component in a narrow-band span of frequencies of the signal. The feedback control block may then further analyze the magnitude and frequency of the first, second, and third dominant spectral components to determine the presence of feedback characteristics in the 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: According to one embodiment, an acoustic signal compensator includes an acoustic signal receiving module, a compensator, and an output module. The acoustic signal receiving module receives an acoustic signal. The compensator performs compensation on the acoustic signal, as compensation of acoustic characteristics of an ear including an ear canal having a first-order resonance characteristic and a second-order resonance characteristic, to suppress a first-order frequency of ear resonance and a second-order frequency lower than double of the first-order frequency. The output module outputs the acoustic signal compensated by the compensator.