Abstract: At least two closely spaced identical or similar loudspeaker assemblies in a horizontal linear array, each loudspeaker assembly comprising at least two identical or similar loudspeakers pointing in different directions so that the loudspeaker assemblies have adjustable, controllable or steerable directivity characteristics. For example, a control module may drive, adjust, control, or steer the loudspeaker assemblies so that at least one acoustic wave field is generated at least at one listening position.

Abstract: A hearing device includes: a first microphone configured to provide of a first microphone input signal; a sound impulse detector configured to detect a sound impulse; a processor configured to provide an electrical output signal based on the first microphone input signal; and a receiver configured to provide an audio output signal based on the electrical output signal; wherein the processor is configured to provide the electrical output signal by performing signal processing in a first set of frequency bands; wherein the sound impulse detector is configured to detect the sound impulse based on a second set of frequency bands, and wherein the second set of frequency bands based on which the sound impulse is detected covers a part of the first set of frequency bands.

Abstract: In order for a microphone array to capture sound emanating from a moving object whose exact position is unknown at the time of arrival of the sound signal, a method for controlling the microphone array comprises steps of receiving position information that includes a position (pTR) and a velocity of the moving object from a tracking system, receiving a plurality of microphone signals that comprise sound of a sound event emanating from the moving object from a plurality of microphone capsules, calculating a directional characteristic from the plurality of microphone signals, wherein the directional characteristic is based on beamforming according to the position information and wherein an audio output signal is generated that includes the sound from a preferred direction of high sensitivity, and providing the audio output signal at an output.

Abstract: An acoustic signal is separated based on a difference in the distance from a sound source to a microphone. By using a filter obtained by associating a value corresponding to an estimated value of a short-distance acoustic signal which is obtained by using “a predetermined function” from a second acoustic signal derived from signals collected by “a plurality of microphones” and is emitted from a position close to “the plurality of microphones” with a value corresponding to an estimated value of a long-distance acoustic signal which is emitted from a position far from “the plurality of microphones”, a desired acoustic signal representing at least one of a sound emitted from a position close to “a specific microphone” and a sound emitted from a position far from “the specific microphone” is acquired from a first acoustic signal derived from a signal collected by “the specific microphone”.

Abstract: Techniques of source localization and acquisition involve a wideband joint acoustic source localization and acquisition approach in light of sparse optimization framework based on an orthogonal matching pursuit-based grid-shift procedure. Along these lines, a specific grid structure is constructed with the same number of grid points as compared to the on-grid case, but which is “shifted” across the acoustic scene. More specifically, it is expected that each source will be located close to a grid point in at least one of the set of shifted grids. The sparse solutions corresponding to the set of shifted grids are combined to obtain the source location estimates. The estimated source positions are used as side information to obtain the original source signals.

Abstract: Endfire linear array microphone systems and methods with consistent directionality and performance at different frequency ranges are provided. The endfire linear array microphone includes a delay and sum beamformer and a differential beamformer. The delay and sum beamformer may produce pickup patterns with good directionality at higher frequency ranges, but cause the pickup patterns to become more omnidirectional at lower frequencies. The differential beamformer may produce pickup patterns with good directionality at lower frequencies. By combining the delay and sum beamformer and differential beamformer within the linear array microphone, the overall directionality of the linear array microphone may be maintained at different frequency ranges while using the same microphone elements.

Abstract: Embodiments include an array microphone comprising a plurality of microphone sets arranged in a linear pattern relative to a first axis and configured to cover a plurality of frequency bands. Each microphone set comprises a first microphone arranged along the first axis and a second microphone arranged along a second axis orthogonal to the first microphone, wherein a distance between adjacent microphones along the first axis is selected from a first group consisting of whole number multiples of a first value, and within each element, a distance between the first and second microphones along the second axis is selected from a second group consisting of whole number multiples of a second value.

Abstract: An electronic device is provided. The electronic device includes a camera, a plurality of microphones, at least one processor electrically coupled with the camera and the plurality of microphones. The at least one processor may acquire a video signal, based on a designated zoom level via the camera, acquire a plurality of audio signals respectively via the plurality of microphones while acquiring the video signal, identify a first signal characteristic of a first audio signal acquired via a first microphone and a second signal characteristic of a second audio signal acquired via a second microphone among the plurality of microphones, derive a control parameter for signal processing for the first audio signal and the second audio signal, based on the designated zoom level, the first signal characteristic, and the second signal characteristic, and perform audio signal processing including beamforming using the first audio signal and the second audio signal, based on the derived control parameter.

Abstract: This disclosure describes an apparatus and method of an embodiment of an invention that is a conferencing apparatus. This embodiment of the apparatus/system includes a microphone array that further comprises a plurality of microphones where each microphone is configured to sense acoustic waves and the plurality of microphones are oriented to develop a corresponding plurality of microphone signals; a processor, memory, and storage operably coupled to the microphone array, the processor configured to: perform a beamforming operation; perform an acoustic echo cancellation operation; perform a direction of arrival determination; and select one or more of the combined echo cancelled signals.

Abstract: A sound processing device is provided with a correction unit that corrects a sound pickup signal. The sound pickup signal is obtained by picking up a sound with a microphone array. The correction unit corrects the sound pickup signal based on directional information that indicates a direction of the microphone array in spherical coordinates, during the picking up of the sound.

Abstract: According to an example embodiment, a technique for spatial audio processing on basis of two or more input audio signals that represent an audio scene and at least one further input audio signal that represents at least part of the audio scene is provided, the technique including identifying a portion of interest (POI) in the audio scene; processing the two or more input audio signals into a spatial audio signal where the POI in the audio scene is suppressed; generating, on basis of the at least one further input audio signal, a complementary audio signal that represents the POI in the audio scene; and combining the complementary audio signal with the spatial audio signal to create a reconstructed spatial audio signal.

Abstract: Provided is a source localization in an apparatus for performing a source localization, a target sound source enhancement or speech recognition. The source localization method using input signals input from a plurality of microphones, comprising steps of: (a) obtaining a log likelihood function or an auxiliary function under the assumption that a target source signal mixed with noises satisfies a CGMM model; (b) obtaining an equation for estimating parameter values of the log likelihood function or the auxiliary function so that a value of the log likelihood function or the auxiliary function is maximized recursively in each time frame; (c) estimating a covariance matrix recursively in each time frame; and (d) estimating a steering vector recursively by using the estimated covariance matrix, wherein the steering vector of the target sound source is estimated from the input signals.

Abstract: A method and device for determining operation of an autonomous device is disclosed. The method includes receiving pixel data and sound data associated with an environment at an instance of time, wherein the pixel data is received from least an image sensor associated with the autonomous device, and wherein the sound data is received from at least four sound sensors placed in a quadrilateral configuration on the autonomous device. Each quadrant of the pixel data is associated with each of the at least four sound sensors. The sound data received is mapped the to the matrix to identify one or more pixels in the matrix corresponding to the sound data based on a difference in amplitude between a first sound sensor of the at least four sound sensors recording maximum sound amplitude with a plurality of second sound sensors of the at least four sound sensors.

Abstract: A howling suppression apparatus includes: an integration processing part that obtains the maximum value among L values corresponding to n-th frames of L i-th signals, for i=1, 2, . . . , L, L being any integer equal to or greater than 2, the L i-th signals being frequency-domain signals obtained from sound signals collected by multiple microphones; and a howling suppression processing part that performs howling suppression processing on at least any of the L i-th signals using the maximum value.

Abstract: A method, computer program product, and computing system for determining a time delay between a first audio signal received on a first audio detection system and a second audio signal received on a second audio detection system. The first and second audio detection systems are located within a monitored space. The first audio detection system is located with respect to the second audio detection system within the monitored space.

Abstract: A binaural hearing system comprises a first and second hearing aids. The hearing aids each comprises antenna and transceiver circuitry allowing the exchange of audio signals between them. At least one of the hearing aids comprises primary and secondary adaptive 2-channel beamformers each providing a spatially filtered signal based on first and second beamformer-input signals. The primary and secondary 2-channel beamformers are coupled in a cascaded structure. In an embodiment, the spatially filtered signal of the secondary 2-channel beamformer may comprise an estimate of user's own voice. In an embodiment, the spatially filtered signal of the secondary 2-channel beamformer may comprise an estimate of a target signal in the environment. In an embodiment, the inputs to the secondary 2-channel beamformer may be beamformed signals from the first and second hearing aids respectively.

Abstract: A system for presenting audio content to a user. The system comprises one or more microphones coupled to a frame of a headset. The one or more microphones capture sound from a local area. The system further comprises an audio controller integrated into the headset and communicatively coupled to an in-ear device worn by a user. The audio controller identifies one or more sound sources in the local area based on the captured sound. The audio controller further determines a target sound source of the one or more sound sources and determines one or more filters to apply to a sound signal associated with the target sound source in the captured sound. The audio controller further generates an augmented sound signal by applying the one or more filters to the sound signal and provides the augmented sound signal to the in-ear device for presentation to a user.

Abstract: A microphone array includes first and second microphones (Ma, Mb) placed on a first axis (f), a third microphone (Mc) placed on a plane (fg) formed by the first axis and a second axis (g) and at a position other than on the first axis, and a fourth microphone (Md) placed on a third axis (h), and at a position other than on the plane formed by the first and the second axes, and a processing circuit generates signals (Cx, Cy, Cz) having bidirectionality in first, second and third mutually perpendicular directions (x, y, z), and an omnidirectional signal (Cw), based on signals (Ba to Bd) obtained by sound collection by means of the first to fourth microphones. It is possible to generate signals having bidirectionality in mutually perpendicular directions, and an omnidirectional signal, without using special microphones, and without excessive restrictions with regard to the placement of the microphones.

Abstract: Systems, methods, tangible non-transitory computer-readable media, and devices associated with detecting and locating sounds are provided. For example, sound data associated with sounds can be received. The sounds can include source sounds and background sounds received by microphones. Based on the sound data, time differences can be determined. Each of the time differences can include a time difference between receipt of a source sound and receipt of a background sound at each of the microphones respectively. A set of the source sounds can be synchronized based on the time differences. An amplified source sound can be generated based on a combination of the synchronized set of the source sounds. A source location of the source sounds can be determined based on the amplified source sound. Based on the source location, control signals can be generated in order to change actions performed by an autonomous vehicle.

Abstract: A non-transitory computer-readable storage medium storing a program that causes a computer to execute a process, the process includes setting a plurality of speaker regions in different directions; calculating a phase difference in each of a plurality of different frequency bands on the basis of a plurality of sound signals acquired by the plurality of microphones; calculating a representative value of the number of phase differences belonging to each of a plurality of phase difference regions corresponding to each of the plurality of speaker regions on the basis of the calculated phase differences and the set plurality of speaker regions; comparing magnitudes of the calculated representative values; and determining, as a direction in which a speaker exists, a direction of a speaker region corresponding to a phase difference region where the compared representative value is large.

Abstract: A device implementing an automatic speech recognition triggering system includes at least one processor configured to receive first and second audio signals respectively corresponding to first and second microphones of a device. The at least one processor is further configured to generate, based on at least one of the first or second audio signals, a third audio signal corresponding to a voice beam directed to an expected position of a mouth of a user. The at least one processor is further configured to determine whether wind noise is present in at least one of the first, second, or third audio signals. The at least one processor is further configured to, based on determining whether wind noise is present, an audio signal from among the second or third audio signals, for a determination of whether at least one of the first or second audio signals corresponds to the user.

Abstract: An electronic device includes one or more microphones that generate audio signals and a wind noise detection subsystem. The electronic device may also include a wind noise reduction subsystem. The wind noise detection subsystem applies multiple wind noise detection techniques to the set of audio signals to generate corresponding indications of whether wind noise is present. The wind noise detection subsystem determines whether wind noise is present based on the indications generated by each detection technique and generates an overall indication of whether wind noise is present. The wind noise reduction subsystem applies one or more wind noise reduction techniques to the audio signal if wind noise is detected. The wind noise detection and reduction techniques may work in multiple domains (e.g., the time, spatial, and frequency domains).

Abstract: Approaches for detecting and reducing wind noise from audio signals captured at multi-microphone array are described. In aspects, the wind noise detector is constructed from probabilities of speech presence and wind noise presence, which are derives from statistics of the phase differences among the time aligned signals of multi-microphones in separate frequency regions. Wind noise, if detected, is reduced by a gain in frequency domain, which is also a function of the phase difference and its statistics.

Abstract: An example method of operation may include designating sub-regions which collectively provide a defined reception space, receiving audio signals at a controller from the microphone arrays in the defined reception space, configuring the controller with known locations of each of the microphone arrays, assigning each of the sub-regions to at least one of the microphone arrays based on the known locations, and creating beamform tracking configurations for each of the microphone arrays based on their assigned sub-regions.

Abstract: Various embodiments of the present disclosure relate to an apparatus and a method for detecting an error of an audio component in an electronic device. The electronic device includes: at least one sound output device; at least one microphone; and a processor, and the processor is configured to identify a deviation related to a loudness of an audio signal acquired through the at least one microphone, and to determine whether an error related to the at least one microphone occurs, based at least on the deviation. Other embodiments are possible.

Abstract: A device for sound localization includes a spatial feature generator, a voice detector, an angle selector, and an angle retriever. The spatial feature generator generates M spatial feature signals according to signals of N microphones of a microphone array. The voice detector generates at least one voice detection signal according to at least one of the signals of the N microphones. The angle selector outputs a candidate angle signal according to the M spatial feature signals to indicate a candidate direction of sound. The angle retriever generates a sound detection result according to the M spatial feature signals to indicate whether any sound source exists, and then outputs an estimated angle signal indicative of a direction of sound according to the sound detection result, the at least one voice detection signal, and the candidate angle signal.

Abstract: Disclosed examples include: determining periods of loudness in an audio signal by: determining whether a difference between a first amplitude at a first time and a second amplitude at a second time satisfies a loudness threshold; in response to the difference satisfying the loudness threshold, setting a loudness flag to a loud state corresponding to the second time, the second time occurring after the first time; determining whether a third amplitude at a third time after the second time satisfies a quiet threshold; in response to the third amplitude satisfying the quiet threshold, setting the loudness flag to a quiet state corresponding to the third time; and determining periods of loudness for the audio signal based on the states of the loudness flag at the second and third times; and indicating the audio signal as spillover based on the periods of loudness.

Abstract: The present technique relates to an apparatus and a method for video-audio processing, and a program each of which enables a desired object sound to be more simply and accurately separated. A video-audio processing apparatus includes a display control portion configured to cause a video object based on a video signal to be displayed; an object selecting portion configured to select the predetermined video object from the one video object or among a plurality of the video objects; and an extraction portion configured to extract an audio signal of the video object selected by the object selecting portion as an audio object signal.

Abstract: The present disclosure is generally directed to an approach for spatializing audio from a received radio transmission to allow a radio operator to audibly perceive audio from the received radio transmission as if originating from a direction that corresponds to a physical location of the transmitting radio. On the receiving side, also referred to herein as a receive (RX) pipeline, a radio device configured consistent with the present disclosure includes utilizing an orientation tracker, and head related transform functions to generate a binaural representation of an incoming transmission such that audio associated with the same gets spatialized to sound as if coming from a direction corresponding to the transmitting radio. On the transmit side, also referred to herein as the transmit (TX) pipeline, includes utilizing a location sensor (e.g., a time of flight and/or GPS sensor) and encoding scheme such that radio transmissions include associated geographical location data.

Abstract: A hearing device includes: a first microphone configured to provide of a first microphone input signal; a sound impulse detector configured to detect a sound impulse; a processor configured to provide an electrical output signal based on the first microphone input signal; and a receiver configured to provide an audio output signal based on the electrical output signal; wherein the processor is configured to provide the electrical output signal by performing signal processing in a first set of frequency bands; wherein the sound impulse detector is configured to detect the sound impulse based on a second set of frequency bands, and wherein the second set of frequency bands based on which the sound impulse is detected covers a part of the first set of frequency bands.

Abstract: A voice-controlled device includes a beamformer for determining audio data corresponding to one or more directions and a beam selector for selecting in which direction a source of target audio lies. The device determines magnitude spectrums for each beam and for each frequency bin in each beam for each frame of audio data. The device determines frame-by-frame changes in the magnitude and filters the changes to smooth them. The device selects the beam having the greatest smoothed change in magnitude as corresponding to speech.

Abstract: In a case where two microphones are used, sound source direction estimation of a plurality of sound sources can be performed with high accuracy. For this purpose, an inter-microphone phase difference is calculated for every frequency band in a microphone pair including two microphones that are installed apart from each other by a predetermined distance. Furthermore, for every frequency band in the microphone pair, a single sound source mask indicating whether or not a component of the frequency band is a single sound source is calculated. Then, the calculated inter-microphone phase difference and the calculated single sound source mask are input as feature quantities to a multi-label classifier, and a direction label associated with a sound source direction is output to the feature quantities.

Abstract: A personal audio device configured to be worn on the head or body of a user and including a plurality of microphones configured to provide a plurality of separate microphone signals capturing audio from an environment external to the personal audio device, and a processor configured to process a first subset of the plurality of separate microphone signals using a first array processing technique to provide a first array signal, compare the first array signal to a microphone signal from the plurality of separate microphone signals, and select the first array signal or the microphone signal based on the comparison.

Abstract: An acoustic goniometer device may include at least four microphones coupled to a collapsible structure. The device may further include a processor configured to receive at least four sound signals from the at least four microphones and to determine a direction of arrival of a sound event within three dimensions based on a time shift between the at least four sound signals. A method may include receiving at least four sound signals from at least four microphones coupled to a collapsible structure and determining a direction of arrival of a sound event within three dimensions based on a time shift between the at least four sound signals.

Abstract: A network of sensors is utilized to capture and relay data that is relevant to specific types of insurance coverage. Sensors included in the network can be deployed as specialized devices, or can be found in existing consumer products. Behavioral and environmental data collected from the sensor network is used to establish a feedback relationship between an insurance policy holder's behavior and insurance policy pricing. Various types of insurance products can benefit from behavioral and environmental data, including health insurance, dental insurance, disability insurance and automobile insurance. Collected data can be used for other purposes, such as initial risk assessment, risk monitoring, and prevention services.

Abstract: An apparatus or method to allow a user to control an audio processing operation of an internal and/or external microphone(s). The method includes providing a configurable user interface which defines an audio processing operation. The status of the audio processing operation can be defined through interaction with the user interface. Capture of sound with the microphone(s) may be controlled based on the status of the audio processing operation.

Abstract: A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V2. The two virtual microphones may be paired with an adaptive filter algorithm and VAD algorithm to significantly reduce the noise without distorting the speech, significantly improving the SNR of the desired speech over conventional noise suppression systems.

Abstract: A system and method is described for determining whether a loudspeaker device has relocated, tilted, rotated, or changed environment such that one or more parameters for driving the loudspeaker may be modified and/or a complete reconfiguration of the loudspeaker system may be performed. In one embodiment, the system may include a set of sensors. The sensors provide readings that are analyzed to determine 1) whether the loudspeaker has moved since a previous analysis and/or 2) a distance of movement and/or a degree change in orientation of the loudspeaker since the previous analysis. Upon determining the level of movement is below a threshold value, the system adjusts previous parameters used to drive one or more of the loudspeakers. By adjusting previous parameters instead of performing a complete recalibration, the system provides a more efficient technique for ensuring that the loudspeakers continue to produce accurate sound for the listener.

Abstract: The disclosure includes a camera array comprising camera modules, the camera modules comprising a master camera that includes a processor, a memory, a sensor, a lens, a status indicator, and a switch, the switch configured to instruct each of the camera modules to initiate a start operation to start recording video data using the lens and the sensor in the other camera modules and the switch configured to instruct each of the camera modules to initiate a stop operation to stop recording, the status indicator configured to indicate a status of at least one of the camera modules. Lens distortion effects may be removed from the frames described by the video data. The camera modules of the camera array are configured to provide a 3× field of view overlap.

Abstract: A microphone apparatus is provided. The microphone apparatus includes a microphone array and an integrated circuit. The microphone array includes at least three microphones arranged in a straight line with a non-uniform configuration. The integrated circuit is electrically connected to the microphone array. The integrated circuit is configured to process a merged sound signal from different combinations of microphones using a time-domain filter to generate an output sound signal. The sound gain of the output sound signal at each sound-receiving angle of each frequency is substantially uniform.

Abstract: A system configured to perform directional speech separation using three or more microphones. The system may dynamically associate direction-of-arrivals with one or more audio sources in order to generate output audio data that separates each of the audio sources. Using three or more microphones, the system may separate audio sources covering 360 degrees surrounding the microphone array, whereas a two-microphone implementation is limited to 180 degrees. The system identifies a target direction for each audio source, dynamically determines directions that are correlated with the target direction, and generates output signals for each audio source. The system may associate individual frequency bands with specific directions based on a phase difference detected by two or more microphones.

Abstract: The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length La, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.

Abstract: The present disclosure generally relates to interfaces and techniques for media playback on one or more devices. In accordance with some embodiments, an electronic device includes a display, one or more processors, and memory. The electronic device receives user input and, in response to receiving the user input, displays, on the display, a multi-device interface that includes: one or more indicators associated with a plurality of available playback devices that are connected to the device and available to initiate playback of media from the device, and a media playback status of the plurality of available playback devices.

Abstract: A microphone array-based target voice acquisition method and device, said method comprising: receiving voice signals acquired on the basis of a microphone array (101); determining a pre-selected target voice signal and a direction thereof (102); performing strong directional gain and weak directional gain on the pre-selected target voice signal, so as to obtain a strong gain signal and a weak gain signal (103); performing an endpoint detection on the basis of the strong gain signal, so as to obtain an endpoint detection result (104); and performing endpoint processing on the weak gain signal according to the endpoint detection result, so as to obtain a final target voice signal (105). The present invention can obtain an accurate and reliable target voice signal, thereby avoiding an adverse effect of the target voice quality on subsequent target voice processing.

Abstract: A microphone array position estimation device includes an estimation unit that estimates a position X of a microphone array for maximizing a simultaneous probability P(X,S,Z) of X, Y, and Z through repeated estimation of S and X when the position of the microphone array constituted by M (M is an integer of 1 or greater) microphones is set to X (=(X1T, . . . , XMT)T, T indicates a transposition), spectrums of sound source signals output by the N (N is an integer of 1 or greater) sound sources are set to S (a set related to all of n, f, and t of Snft, f is a frequency bin, and t is a frame index), and spectrums of recorded signals collected by the microphone array are set to Z (a set related to all of f and t of Zft).

Abstract: In one embodiment, a method for matching sensors includes generating a first and second sensor signals respectively from first and second sensors, separating the sensor signals into magnitude and phase components, determining a phase difference from the phase components, and matching the magnitude of the first sensor signal to that of the second sensor signal by multiplying the magnitude of the first sensor signal by a magnitude correction value that is a function of a ratio of the phase components of the first and second sensor signals.

Abstract: An example method of operation may include receiving audio data, via one or more microphones of a network device, from an audio source, determining a location of the audio data based on a direction and amplitude of the received audio data, modifying the audio data for output via a loudspeaker of the network device, and outputting, via the loudspeaker, the modified audio data.

Abstract: The present disclosure provides a transistor acoustic sensor element and a method for manufacturing the same, an acoustic sensor and a portable device. The transistor acoustic sensor element comprises a gate, a gate insulating layer, a first electrode, an active layer and a second electrode arranged on a base substrate, wherein the active layer has a nanowire three-dimensional mesh structure and thus can vibrate under the action of sound signals, so that the output current of the transistor acoustic sensor element changes correspondingly. Since the active layer having the nanowire three-dimensional mesh structure can sensitively sense weak vibration of acoustic waves, the sensitivity to sound signals of the transistor acoustic sensor element is improved.

Abstract: Methods and systems for performing modal reverb techniques for audio signals are described. The method may involve simplifying a reverb effect to be applied to the audio signal by receiving an IR, dividing the IR into a plurality of sub-bands, using a parametric estimation algorithm to determine respective parameters of the modes included in each sub-band, aggregating the respective modes of the sub-bands into a set; and truncating the set of aggregated modes into a subset of modes. Reverberation of the audio signal may be manipulated based on an IR that itself is based on the truncated subset of modes.

Abstract: There is provided a signal processing apparatus that obtains a speech signal with sufficiently reduced wind noise.