Abstract: A system comprising at least one processor; and at least one storage device. The storage device(s) store instructions that, when executed, cause the at least one processor to: prior to enabling output of an audio signal based on an audio data stream, detect, within the audio data stream, an indication of a target sound that corresponds to one of a plurality of sounds that are expected to cause distraction, replace, within the audio data stream, the indication of the target sound with an indication of a replacement sound, wherein the replacement sound is a less distracting version of the target sound, and after replacing the indication of the target sound with the indication of the replacement sound, output the audio data stream.

Abstract: A method performs noise suppression in hearing aids. A step of the method involves a first audio signal being split into a plurality of essentially disjunct frequency bands. A further step of the method involves a reference band being selected from the plurality of frequency bands, which reference band has an establishable first component of a speech signal. Another step involves a correlation between the reference band and a first frequency band being ascertained. A further step involves a value that indicates a second component of a speech signal in the first frequency band being ascertained on the basis of the correlation. Another step of the method involves a noise suppression being set in the first frequency band on the basis of the ascertained value.

Abstract: A method for adjusting a degree of filtering applied to an audio signal includes modeling a probability density function (PDF) of a fast Fourier transform (FFT) coefficient of a primary channel and reference channel of the audio signal; maximizing at least one of PDFs to provide a discriminative relevance difference (DRD) between a noise magnitude estimate of the reference channel and a noise magnitude estimate of the primary channel. The method further includes emphasizing the primary channel when the spectral magnitude of the primary channel is stronger than the spectral magnitude of the reference channel; and deemphasizing the primary channel when the spectral magnitude of the reference channel is stronger than the spectral magnitude of the primary channel.

Abstract: There is provided a method and device for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A set of local maxima of a cross-correlation function involving at least two different channels of the multi-channel audio signal is determined (S1) for positive and negative time-lags, where each local maximum is associated with a corresponding time-lag. From the set of local maxima, a local maximum for positive time-lags is selected as a so-called positive time-lag inter-channel correlation candidate and a local maximum for negative time-lags is selected as a so-called negative time-lag inter-channel correlation candidate (S2). When the absolute value of a difference in amplitude between the inter-channel correlation candidates is smaller than a first threshold, it is evaluated whether there is an energy-dominant channel (S3).

Abstract: A UAV is provided to cancel background noise from audio data collected by the UAV. The UAV is provided with one or more background microphones in a proximity of one or more background noise-producing components. The UAV is also provided with one or more audio source collecting microphones. The audio data collected by the background microphones may be used to reduce or cancel interfering background noise from the audio signal detected by the audio source collecting microphone. The target audio may be captured or recorded with little or no background noise.

Abstract: An audio playback system generates output signals for multiple channels of acoustic transducers by applying a rendering matrix to data representing the aural content and spatial characteristics of audio objects, so that the resulting sound field creates accurate listener impressions of the spatial characteristics. Matrix coefficients are updated to render moving objects. Discontinuous updates of the rendering matrix coefficients are controlled according to psychoacoustic principles to reduce audible artifacts. The updates may also be managed to control the amount of data needed to perform the updates.

Abstract: A system for self-organized acoustic signal cancellation over a network is disclosed. The system may transmit an acoustic sounding signal to an interfering device so that a channel measurement may be performed for a channel between the interfering device and an interferee device. The system may receive the channel measurement for the channel from the interfering device and also receive a digitized version of an audio interference signal associated with the interfering device. Based on the channel measurement and the digital version of the interference signal, the system may calculate a cancellation signal prior to the arrival of the original over-the-air audio interference signal that corresponds to the digital version of audio interference signal. The system may then apply the cancellation signal to an audio signal associated with the interferee device to remove the interference signal from the audio signal.

Abstract: A method includes determining an error condition during a bandwidth transition period of an encoded audio signal. The error condition corresponds to a second frame of the encoded audio signal, where the second frame sequentially follows a first frame in the encoded audio signal. The method also includes generating audio data corresponding to a first frequency band of the second frame based on audio data corresponding to the first frequency band of the first frame. The method further includes re-using a signal corresponding to a second frequency band of the first frame to synthesize audio data corresponding to the second frequency band of the second frame.

Abstract: Methods and systems are provided for separating a target speech from a plurality of other speeches having different directions of arrival. One of the methods includes obtaining speech signals from speech input devices disposed apart in predetermined distances from one another, calculating a direction of arrival of target speeches and directions of arrival of other speeches other than the target speeches for each of at least one pair of speech input devices, calculating an aliasing metric, wherein the aliasing metric indicates which frequency band of speeches is susceptible to spatial aliasing, enhancing speech signals arrived from the direction of arrival of the target speech signals, based on the speech signals and the direction of arrival of the target speeches, to generate the enhanced speech signals, reading a probability model, and inputting the enhanced speech signals and the aliasing metric to the probability model to output target speeches.

Abstract: A planning system (201) for scheduling the operation of autonomous entities within a defined geographical region. The planning system operates at a region plan level (301) for strategic planning across the geographical region, at an operation plan level (302) for operations to be performed by autonomous entities in localized zones having operation-defined geographical boundaries, and at a task plan level (303) in which processing is undertaken in respect of specific tasks to be performed by the autonomous entities, in undertaking the operations.

Abstract: A first control signal filter to which a cosine wave oscillating at a control frequency is input; a second control signal filter to which a sine wave oscillating at the control frequency is input; a control signal adder for outputting a control signal generated by adding an output of the first control signal filter and an output of the second control signal filter; a filter coefficient update unit for updating filter coefficients of the first control signal filter and the second control signal filter; and a frequency correction value calculation unit for calculating a frequency correction value for correcting the control frequency on the basis of the control signal and the control frequency.

Abstract: Acoustic transducer aging compensation is effective for an acoustic transducer that is driven with an adjustable drive power to output a signal. A microphone can measure the amplitude of the transmitted signal corresponding to a transmitted sound pressure level (SPL). A controller can periodically compare the transmitted SPL to the drive power or a previous SPL, and determine if the received SPL has declined with respect to the input drive power over time, whereupon the controller can direct an increase in drive power to the SPL-declined acoustic transducer to compensate for the decline in received SPL. If drive power is at a maximum, the controller can further instruct a mobile device receiver to lower its receiver detection threshold for the signal from the SPL-declined acoustic transducer to further compensate for the decline in SPL from that acoustic transducer.

Abstract: A voice processing device includes a noise-originating coefficient calculation section that calculates a noise-originating coefficient that gradually decreases as a target value of stationary noise for each frequency increases, the target value being calculated based on an amplitude value of a frequency spectrum obtained by time-frequency transforming a voice signal for a predetermined period of time, and a suppression signal generation section that generates, when the frequency spectrum is determined as being stationary on the basis of the amplitude value, a suppression signal by multiplying a suppression coefficient based on the noise-originating coefficient by the amplitude value, the suppression signal being frequency-time transformed to be output.

Abstract: At least one signal is received that represents speech and noise. In response to the at least one signal, frequency bands are generated of an output channel that represents the speech while attenuating at least some of the noise from the at least one signal. Within a kth frequency band of the at least one signal: a first ratio is determined of a clean version of the speech for a preceding time frame to the noise for the preceding time frame; and a second ratio is determined of a noisy version of the speech for the time frame n to the noise for the time frame n. In response to the first and second ratios, a gain is determined for the kth frequency band of the output channel for the time frame n.

Abstract: Methods and systems are provided for separating a target speech from a plurality of other speeches having different directions of arrival. One of the methods includes obtaining speech signals from speech input devices disposed apart in predetermined distances from one another, calculating a direction of arrival of target speeches and directions of arrival of other speeches other than the target speeches for each of at least one pair of speech input devices, calculating an aliasing metric, wherein the aliasing metric indicates which frequency band of speeches is susceptible to spatial aliasing, enhancing speech signals arrived from the direction of arrival of the target speech signals, based on the speech signals and the direction of arrival of the target speeches, to generate the enhanced speech signals, reading a probability model, and inputting the enhanced speech signals and the aliasing metric to the probability model to output target speeches.

Abstract: A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.

Abstract: Methods and systems are provided for implementing a distributed algorithm for beam-forming (e.g., MVDR beam-forming) using a message-passing algorithm. The message-passing algorithm provides for computations to be performed in a distributed manner across a network, rather than in a centralized processing center or “fusion center”. The message-passing algorithm may also function for any network topology, and may continue operations when various changes are made in the network (e.g., nodes appearing, nodes disappearing, etc.). Additionally, the message-passing algorithm may minimize the transmission power per iteration and, depending on the particular network, also may minimize the transmission power required for communication between network nodes.

Abstract: In aspects of concurrent sound source localization of multiple speakers, audio signals from two or more microphones are upsampled, and then the upsampled audio signals are time-multiplexed to a plurality of beamformers. A first sound source received at the two or more microphones is localized at a first beamformer, and a second sound source received at the two or more microphones is localized at a second beamformer, where localizing the second sound source is constrained by the localization of the first sound source. The beamformers can filter the upsampled audio signals using beamformer coefficients from the localizations to produce beamformed audio signals.

Abstract: Provided are systems and methods for generating clean speech from a speech signal representing a mixture of a noise and speech. The clean speech may be generated from synthetic speech parameters. The synthetic speech parameters are derived based on the speech signal components and a model of speech using auditory and speech production principles. The modeling may utilize a source-filter structure of the speech signal. One or more spectral analyzes on the speech signal are performed to generate spectral representations. The feature data is derived based on a spectral representation. The features corresponding to the target speech according to a model of speech are grouped and separated from the feature data. The synthetic speech parameters, including spectral envelope, pitch data and voice classification data are generated based on features corresponding to the target speech.

Abstract: A system for providing an acoustic environment recognizer for optimal speech processing is disclosed. In particular, the system may utilize metadata obtained from various acoustic environments to assist in suppressing ambient noise interfering with a desired audio signal. In order to do so, the system may receive an audio stream including an audio signal associated with a user and including ambient noise obtained from an acoustic environment of the user. The system may obtain first metadata associated with the ambient noise, and may determine if the first metadata corresponds to second metadata in a profile for the acoustic environment. If the first metadata corresponds to the second metadata, the system may select a processing scheme for suppressing the ambient noise from the audio stream, and process the audio stream using the processing scheme. Once the audio stream is processed, the system may provide the audio stream to a destination.

Abstract: An apparatus for eliminating noise includes: a gain acquisition unit that determines a gain and a correction value of the gain using a signal to noise ratio (SNR) of an input signal; and a gain application unit that acquires an output signal corresponding to the input signal using the determined gain and the determined correction value, wherein the output signal includes an input signal of which noise is eliminated and an input signal of which noise is not eliminated, and a proportion of the input signal of which noise is eliminated and a proportion of the input signal of which noise is not eliminated are determined according to the determined correction value.

Abstract: In some embodiments, a method for monitoring speakers within an audio playback system (e.g., movie theater) environment. In typical embodiments, the monitoring method assumes that initial characteristics of the speakers (e.g., a room response for each of the speakers) have been determined at an initial time, and relies on one or more microphones positioned in the environment to perform a status check on each of the speakers to identify whether a change to at least one characteristic of any of the speakers has occurred since the initial time. In other embodiments, the method processes data indicative of output of a microphone to monitor audience reaction to an audiovisual program. Other aspects include a system configured (e.g., programmed) to perform any embodiment of the inventive method, and a computer readable medium (e.g., a disc) which stores code for implementing any embodiment of the inventive method.

Abstract: An approach to processing of acoustic signals acquired at a user's device include one or both of acquisition of parallel signals from a set of closely spaced microphones, and use of a multi-tier computing approach in which some processing is performed at the user's device and further processing is performed at one or more server computers in communication with the user's device. The acquired signals are processed using time versus frequency estimates of both energy content as well as direction of arrival. In some examples, a non-negative matrix or tensor factorization approach is used to identify multiple sources each associated with a corresponding direction of arrival of a signal from that source. In some examples, data characterizing direction of arrival information is passed from the user's device to a server computer where direction-based processing is performed.

Abstract: A method, system, and computer program product for processing an encoded audio signal is described. In one exemplary embodiment, the system receives an encoded low-frequency range signal and encoded energy information used to frequency shift the encoded low-frequency range signal. The low-frequency range signal is decoded and an energy depression of the decoded signal is smoothed. The smoothed low-frequency range signal is frequency shifted to generate a high-frequency range signal. The low-frequency range signal and high-frequency range signal are then combined and outputted.

Abstract: Loudness control is performed by estimating the energy and loudness of an audio signal. Loudness of the audio signal is determined by decomposing the audio signal into multiple frequency bands at different center frequencies. The energy of the audio signal in each frequency band is calculated and converted to a loudness, using a function that models human loudness perceptions of audio energies at different frequencies. The loudnesses are summed to obtain the total loudness of the audio signal. A signal gain is calculated as a function of a loudness setting provided by a human listener, the estimated signal energy, and the total loudness of the audio signal.

Abstract: A sound outputting apparatus and a method of controlling the same are provided. A method of controlling a sound outputting apparatus includes extracting a noise signal and a desired signal, estimating a direction of arrival (DoA) of the extracted desired signal, and outputting sound.

Abstract: A system for self-organized acoustic signal cancellation over a network is disclosed. The system may transmit an acoustic sounding signal to an interfering device so that a channel measurement may be performed for a channel between the interfering device and an interferee device. The system may receive the channel measurement for the channel from the interfering device and also receive a digitized version of an audio interference signal associated with the interfering device. Based on the channel measurement and the digital version of the interference signal, the system may calculate a cancellation signal prior to the arrival of the original over-the-air audio interference signal that corresponds to the digital version of audio interference signal. The system may then apply the cancellation signal to an audio signal associated with the interferee device to remove the interference signal from the audio signal.

Abstract: A robust noise suppression system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to cochlea domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: According to one embodiment, an apparatus for discriminating speech/non-speech of a first acoustic signal includes a weight assignment unit, a feature extraction unit, and a speech/non-speech discrimination unit. The weight assignment unit is configured to assign a weight to each frequency band, based on a frequency spectrum of the first acoustic signal including a user's speech and a frequency spectrum of a second acoustic signal including a disturbance sound. The feature extraction unit is configured to extract a feature from the frequency spectrum of the first acoustic signal, based on the weight of each frequency band. The speech/non-speech discrimination unit is configured to discriminate speech/non-speech of the first acoustic signal, based on the feature.

Abstract: To provide a noise suppressing method and apparatus capable of achieving high-quality noise suppression using a lower amount of operations. Noise contained in an input signal is suppressed by transforming the input signal into frequency-domain signals; integrating bands of the frequency-domain signals to determine integrated frequency-domain signals; determining estimated noise based on the integrated frequency-domain signals; determining spectral gains based on the estimated noise and said integrated frequency-domain signals; and weighting said frequency-domain signals by the spectral gains.

Abstract: A noise reduction device may be provided. The noise reduction device may include: an input configured to receive an input signal including a representation in a frequency domain of an audio signal, wherein the representation includes a plurality of time frames and a plurality of coefficients for each time frame; a noise detection circuit configured to determine a first indicator being indicative of a bandwidth of a coefficient over at least two time; a noise reduction circuit configured to reduce based on the first indicator a noise component in the audio signal; and an output configured to output an output signal including a representation in the frequency domain of the audio signal with the reduced noise component.

Abstract: A method for estimating acoustic noise in an environment where a mobile communication device is operating and where the acoustic noise includes nonstationary noise or speech-like noises, and wherein the environment also includes speech signals. The method includes searching for a local minimum energy over a plurality of frames using at least two reference signals including a first signal comprised of a time-sensitive current local minimum energy estimate, emin, and a second signal comprised of a time-weighted average of previous detected local energy minima, eminmean; and deciding whether the detected local energy minima of the second reference signal is a noise signal. Also, binning the detected input signal energy minima values within a plurality of histograms; and calculating a composite noise energy estimate comprised of a weighted sum of a maximum probability noise energy estimate and an expected value noise energy estimate. As such a nonstationary noise estimator is formed.

Abstract: An embodiment of the invention provides a noise cancellation method for an electronic device. The method comprises: receiving an audio signal; applying a Fast Fourier Transform operation on the audio signal to generate a sound spectrum; acquiring a first spectrum corresponding to a noise and a second spectrum corresponding to a human voice signal from the sound spectrum; estimating a center frequency according to the first spectrum and the second spectrum; and applying a high pass filtering operation to the sound spectrum according to the center frequency.

Abstract: A noise suppression apparatus selectively uses an adaptive beamformer and fixed beamformer for each frequency. A direction of a null of the fixed beamformer is determined from a direction of a null automatically formed by the adaptive beamformer. Filter coefficients of the adaptive beamformer based on an output power minimization rule are calculated by a minimum norm method using a norm of the filter coefficients as a constraint. The above selection is made based on, for example, a depth of a null automatically formed by the adaptive beamformer in the selection.

Abstract: Limiting the sound pressure level presented to the listener's ears by one or more headphones, using processing capabilities of a personal media device. Headphones, coupled to audio signals from a personal media device, include a sensor to measure the sound pressure level presented to the listener's ears, and provide that measure to the personal media device. The personal media device, optionally aided by one or more analog circuits, adjusts the audio signal so that the sound pressure level is maintained within a recommended range.

Abstract: Provided is a noise suppressing technology capable of suppressing various noises including unknown noises without storing information relating to a large number of noises in advance. Noises in a degraded signal are suppressed and noise information is generated on the basis of a noise suppression result. The noises in the degraded signal are suppressed using the generated noise information.

Abstract: Audio processing devices and methods. An ambient microphone picks up ambient sound and a voice microphone picks up a speaker's voice. The voice microphone is located farther from the speaker's mouth when a device is held to one side of the speaker's face than when held to the other side. A signal indicates whether an expected position of the handheld body is a position on a left side or right side of a speaker's face. Ambient sound in the voice signal is reduced by applying a first algorithm configuration selected to process voice signals with lower signal-to-noise ratio when the expected position in on one side of the speaker's face, and by applying a second sound cancellation algorithm configuration selected to process higher signal-to-noise ratios when the expected position is on the other side of the speaker's face.

Abstract: A method and device for adaptively adjusting sound effect, and the method comprises: obtaining an energy value of the current ambient noise; receiving a first trigger instruction and adjusting the current output volume based on the energy value of the current ambient noise; while judging that the energy value of the current ambient noise is bigger than a first threshold, processing treble enhancement; while judging that the energy value of the current ambient noise is less than a second a sound threshold, processing bass enhancement. By collecting the voice data and detecting the speech activity on the voice data, when the first trigger instruction is received, the method can adjust the current volume and adjust the frequency response by the treble enhancement or the bass enhancement based on the energy value of the current ambient noise, thereby obtaining the better sound effect and easy to achieve.

Abstract: A noise canceling system comprises a microphone (103) for generating a captured signal representing sound in an audio environment and a sound transducer (101) for radiating a sound canceling audio signal in the audio environment. A feedback path (105, 107, 109, 111, 113) exists from the microphone (103) to the sound transducer (101) and comprises a feedback filter (109). A tone processor (119) determines a tone component characteristic for a tone component of a feedback signal of the feedback path (105, 107, 109, 111, 113) and an adaptation processor (121) adapts the feedback path in response to the tone component characteristic. The invention allows detection of the onset of instability and dynamic compensation to mitigate or prevent such instability. Accordingly increased design freedom for the feedback filter is achieved resulting in improved noise cancellation.

Abstract: A method and apparatus for enhancing a desired audio signal for delivery through an electroacoustic channel include obtaining a noise estimate attributable to an external disturbance, applying the noise estimate to a dynamic noise compensation (DNC) process to thereby condition the desired audio signal as a function of the spectral characteristics of the noise estimate, applying the noise estimate to an adaptive equalization (AEQ) process to thereby condition the desired audio signal as a function of the electroacoustic response of the electroacoustic channel, and applying the noise estimate to an active noise cancellation (ANC) process configured to generate anti-noise for delivery into the electroacoustic channel.

Abstract: A signal processing device processes a plurality of observed signals at a plurality of frequencies. The plurality of the observed signals are produced by a plurality of sound receiving devices which receive a mixture of a plurality of sounds. In the signal processing device, a storage stores observed data of the plurality of the observed signals. The observed data represents a time series of magnitude of each frequency in each of the plurality of the observed signals. An index calculator calculates an index value from the observed data for each of the plurality of the frequencies. The index value indicates significance of learning of a separation matrix using the observed data of each frequency. The separation matrix is used for separation of the plurality of the sounds from each other at each frequency. A frequency selector selects one or more frequency according to the index value of each frequency.

Abstract: A mobile device includes a processing device that converts a digital input signal to an analog output signal and a memory device that stores a plurality of gain tables, each including values associated with processing the digital input signal. The mobile device further includes at least one filter and at least one amplifier. The processing device is configured to select one of the plurality of gain tables based on the signal strength of the digital input signal and apply one or more values in the selected gain table to the filter, the amplifier, or both, to process the digital input signal, the analog output signal, or both. A method includes receiving a digital input signal, identifying a signal strength of the digital input signal, selecting one of a plurality of gain tables, and processing the digital input signal based on the selected gain table.

Abstract: A vehicular microphone system (200) for post processing optimization of a microphone signal includes a first transducer (201) and second transducer (203) separated by a predetermined distance within an automotive mirror. A first high pass filter network (205) is connected to the first transducer (201) while a second high pass filter network (207) connected to the second transducer (203). A low frequency shelving filter (209) is used for receiving the output from the second high pass filter (207). A first all pass filter (211) is connected to the low frequency shelving filter (209) and a second all pass filter (213) is used in connection with the first all pass filter (211) for tailoring audio characteristics.

Abstract: A noise removal unit 102 executes noise removal and flooring processing of an input signal, and a density calculating unit 104 calculates, as to a point of interest on a time-frequency plane of the input signal passing through the noise removal, a density of non-flooring processing points from the presence or absence of the flooring processing of individual points around the point of interest. A partial suppression unit 105 replaces, when the density is less than a threshold, the power of the point of interest with its flooring value by considering it as a musical noise component, thereby suppressing the musical noise component.

Abstract: An audio beamforming apparatus includes a receiving circuit (103) which receives signals from an at least two-dimensional microphone array (101). A reference circuit (105) generates reference beams and a combining circuit (107) generates an output signal corresponding to a desired beam pattern by combining the reference beams. An estimation circuit (109) generates a direction estimate by determining angles corresponding to local minima for a power measure of the output signal in at least a first and respectively second angle interval. The direction estimate is generated by selecting one of the angles. The combining circuit (107) determines combination parameters to provide a notch in an angle corresponding to the direction estimate and a maximization of a directivity cost measure where the directivity cost measure is indicative of a ratio between a gain in the first direction and an energy averaged gain.

Abstract: A system and method for reducing uplink noise in a mobile communications device, the system including: a noise estimator for estimating noise in proximity to the mobile communication device; an adjustable filter for receiving a signal from a microphone of the mobile communication device; an adjustable attenuation block for receiving a filtered signal from the adjustable filter; a controller configured to: monitor the estimated noise; and adjust the adjustable filter and adjustable attenuation block based on the estimated noise. In particular, the controller may be configured to adjust the adjustable filter by increasing the depth of the filtering for higher estimated noise levels and adjust the attenuation by increasing the attenuation for higher estimated noise levels.

Abstract: The present invention discloses methods, apparatus and systems for individualizing music, audio and speech adaptively, intelligently and interactively according to a listener's personal hearing ability, unique hearing preference, characteristic feedback, and real-time surrounding environment.

Abstract: Beamformer coefficients may include a plurality of sets of theoretical statistical data for theoretical signals. Each theoretical signal may have its own particular attributes. The statistical data may be used in computing beamformer coefficients for application by a beamformer to signals received at a device. Signals are received at an input of the device. A respective plurality of weights is determined, for the theoretical statistical data sets, based on an analysis of the extent to which the signals have the particular attributes of the theoretical signals. The theoretical are retrieved, and a statistical data set is calculated for the signals by performing a weighted sum of the theoretical statistical data sets using the determined respective plurality of weights. Beamformer coefficients are computed based on the calculated statistical data set for the signals, which are used by a beamformer to the signals for generating a beamformer output.

Abstract: A purpose of the invention is to provide a noise gate that can output an audio signal in which only a stationary noise is removed, without degrading an utterance voice of a speaking person. A sound collection device 1 includes an FFT processing unit 11, the noise gate 12, and an IFFT processing unit 13. The sound collection device 1 transforms a collected audio signal NET into a frequency spectrum NE?N by using the FFT processing unit 11. The noise gate 12 estimates a noise spectrum N?N of a stationary noise based on the frequency spectrum NE?N of the audio signal. The noise gate 12 decreases a signal level (a gain) of the audio signal in a case where a signal level ratio of the frequency spectrum NE?N of the audio signal to the noise spectrum N?N is less than a threshold value, and outputs the audio signal.