Abstract: A binaural decoder for an MPEG surround stream, which decodes an MPEG surround stream into a stereo 3D signal, and a decoding method thereof. The method includes dividing a compressed audio stream and head related transfer function (HRTF) data into subbands, selecting predetermined subbands of the HRTF data divided into subbands and filtering the HRTF data to obtain the selected subbands, decoding the audio stream divided into subbands into a stream of multi-channel audio data with respect to subbands according to spatial additional information, and binaural-synthesizing the HRTF data of the selected subbands with the multi-channel audio data of corresponding subbands.

Abstract: A position locating method and apparatus are applied to the electronic information field, so that shifting an indication cursor in a wide range using a wireless indication device can be avoided, and a user operation can be simplified. The method comprises detecting a line-of-sight orientation of a user, and obtaining a line-of-sight orientation parameter of the user; detecting a line-of-sight source of the user, and obtaining a line-of-sight source position of the user; obtaining a position of a screen, and obtaining a screen position of a line of sight according to the line-of-sight orientation parameter, the position of the screen, and the line-of-sight source position; and receiving an adjustment signal, adjusting, according to the adjustment signal, the position displayed on the screen, and displaying an adjusted position on the screen.

Abstract: An information processing apparatus includes a holding unit configured to hold a plurality of head related transfer functions for outputting directional sound in a plurality of directions, a setting unit configured to set a direction in which a first head related transfer function and a second head related transfer function are switched, based on characteristics of the first head related transfer function and the second head related transfer function, and a switching unit configured to switch a head related transfer function used to output the directional sound between the first head related transfer function and the second head related transfer function in the set direction.

Abstract: Provided are an audio signal processing method and apparatus for adjusting a location of an audio object in correspondence to a location of a visual object. The audio signal processing apparatus includes a matching unit configured to select an audio object corresponding to a visual object extracted from a video signal among at least one audio object extracted from an audio signal, a location adjusting unit configured to adjust a location of a sound image of the audio signal based on a location of the selected audio object and a location of a visual object corresponding to the selected audio, and an output unit configured to output an audio signal whose the location of the sound image is adjusted.

Abstract: An audio providing apparatus and method are provided. The audio providing apparatus includes: an object renderer configured to render an object audio signal based on geometric information regarding the object audio signal; a channel renderer configured to render an audio signal having a first channel number into an audio signal having a second channel number; and a mixer configured to mix the rendered object audio signal with the audio signal having the second channel number.

Abstract: Provided herein are a headphone amplifier circuit for a headphone driver, an operation method thereof, and a universal serial bus (USB) interfaced headphone device using the same. The output stage of the headphone amplifier circuit is improved to have a differential output structure so as to effectively solve the crosstalk issue and increase the isolation between the left and the right channels such that the audio content presents a spatial perception and a distance perception more specifically.

Abstract: A directivity control apparatus is provided which controls a directivity of a sound collected by a first sound collector including a plurality of microphones. The directivity control apparatus forms a directivity of the sound in a direction toward a position of a monitoring target in an image displayed on a display. Information on the position of the monitoring target in the image displayed on the display is obtained. The directivity of the sound is changed toward the position of the monitoring target in accordance with a movement of the monitoring target by referring to the obtained information on the position of the monitoring target.

Abstract: Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics (HOA) uses spherical harmonics of at least 2nd order. However, commonly used loudspeaker setups are irregular and lead to problems in decoder design. A method for improved decoding an audio soundfield representation for audio playback comprises calculating a panning function (W) using a geometrical method based on the positions of a plurality of loudspeakers and a plurality of source directions, calculating a mode matrix (?) from the loudspeaker positions, calculating a pseudo-inverse mode matrix (?+) and decoding the audio soundfield representation. The decoding is based on a decode matrix (D) that is obtained from the panning function (W) and the pseudo-inverse mode matrix (?+).

Abstract: An electronic device receives a first distance signal from a sensor embraced by a left receiver of a headset, and a second distance signal from a sensor embraced by a right receiver of the headset. The electronic device obtains a first distance between the left receiver and a left ear of the person according to the first distance signal, and obtains a second distance between the right receiver and a right ear of the person according to the second distance signal. The electronic device executes a specific operation according to the first distance and the second distance.

Abstract: A system for monitoring two or more persons in an area with an entry point having an access allowed indication or an access denied indication. a token device in communication with at least one wireless proximity detection device which determines a relative location of the token device to provide a detection data point for each of the wireless proximity detection devices and a set of detection data points for the group of detection data points; a system computing device which calculates the shared proximity of the token device according to the set of detection data points and determines that the token device contains at least two valid tickets or does not, if there are valid tickets and the shared proximity of the token device is within a predetermined area the access allowed indication will display to allow entry for a number of persons corresponding to the number of the tickets.

Abstract: A system and method for joint acoustic echo control and adaptive array processing, comprising the decomposition of a captured sound field into N sub-sound fields, applying linear echo cancellation to each sub-sound field, selecting L sub-sound fields from the N sub-sound fields, performing L channel adaptive array processing utilizing the L selected sub-sound fields, and applying non-linear audio echo cancellation.

Abstract: A domain description is received, by a processor, the domain description identifying a domain associated with a sensor input. The domain description is formatted according to a hierarchical naming structure. A training data set is selected from a plurality of training data sets based upon the received domain description and sensor input. A combination of a subset of classifiers for classifying the sensor input is selected from a set of classifiers based upon the selected training data set.

Abstract: Disclosed herein are system, method, and tangible computer readable medium for creating a desired audio effect for a user. The method includes operations including: causing a plurality of speakers to play test signals, each test signal being specific to one of the speakers; receiving from a remote device recorded frequency responses of the speakers resulting from the playing of the test signals; creating one or more filters to match an audio profile selected by a user; applying the filters to the recorded frequency responses to obtain filtered transformations of the speakers; and transmitting the filtered transformations to the speakers; wherein the filtered transformations are applied at the speakers to thereby achieve the user audio profile.

Abstract: A binaural hearing system (“system”) preserves and/or enhances interaural level differences (“ILDs”) between first and second signals. The system includes audio detectors each associated with an ear of a user. The audio detectors detect an audio signal presented to the user and generate the first and second signals to represent the audio signal as detected at each ear. The system also includes sound processors associated with each ear that each receive the first and second signals from the audio detectors directly and/or by way of a communication link from the other sound processor. The sound processors each perform operations with respect to the first and second signals to preserve and/or enhance the ILDs between the signals. In so doing, the sound processors perform a contralateral gain synchronization operation to a first degree at the first sound processor and to a distinct second degree at the second sound processor.

Abstract: An audio decoder device for decoding a compressed input audio signal having at least one core decoder having one or more processors for generating a processor output signal based on a processor input signal, wherein a number of output channels of the processor output signal is higher than a number of input channels of the processor input signal, wherein each of the one or more processors has a decorrelator and a mixer, wherein a core decoder output signal having a plurality of channels has the processor output signal, and wherein the core decoder output signal is suitable for a reference loudspeaker setup; at least one format converter device configured to convert the core decoder output signal into an output audio signal, which is suitable for a target loudspeaker setup; and a control device configured to control at least one or more processors in such way that the decorrelator of the processor may be controlled independently from the mixer of the processor, wherein the control device is configured to contro

Abstract: A method for securing a computer system comprising a step of: disguising a password entry screen.

Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by a new type of decode matrix for sound field data, and a new way to obtain the decode matrix.

Abstract: A system and method are described for transforming stereo signals into mid and side components xm and xs to apply processing to only the side-component xs and avoid processing the mid-component. By avoiding alteration to the mid-component XM, the system and method may reduce the effects of ill-conditioning, such as coloration that may be caused by processing a problematic mid-component xM while still performing crosstalk cancellation and/or generating virtual sound sources. Additional processing may be separately applied to the mid and side components xM and xs and/or particular frequency bands of the original stereo signals to further reduce ill-conditioning.

Abstract: The present invention provides a method of processing a source left view and a source right view of a 3D image comprising the steps of: identifying at least one ghosted region or ghosting region in the left view and/or in the right view; processing the identified regions in order to create a binocular suppression effect for this identified regions then providing a new 3D image formed by the processed left and/or right views.

Abstract: An audio providing apparatus and method are provided. The audio providing apparatus includes: an object renderer configured to render an object audio signal based on geometric information regarding the object audio signal; a channel renderer configured to render an audio signal having a first channel number into an audio signal having a second channel number; and a mixer configured to mix the rendered object audio signal with the audio signal having the second channel number.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: A method and an electronic device for outputting an audio signal in the electronic device is provided. The electronic device includes a first speaker, a second speaker, and an audio processor that creates, from an audio signal, a first frequency audio signal corresponding to a first frequency band by using a low pass filter, synthesizes the created first frequency audio signal and the audio signal to create a synthetic audio signal, creates, from the synthetic audio signal, a second frequency audio signal corresponding to a second frequency band by using a high pass filter, outputs the created second frequency audio signal through the first speaker, and outputs the created synthetic audio signal through the second speaker.

Abstract: Technology for adjusting a receiver timing of a wireless device in a Coordinated MultiPoint (CoMP) system is disclosed. One method can include the wireless device receiving a plurality of node specific reference signals (RSs) from a plurality of cooperating nodes in a coordination set of the CoMP system. The coordination set includes at least two cooperating nodes. The wireless device can estimate a composite received RS timing from a plurality of received RS timings generated from the plurality of node specific RSs. The received RS timings represent timings from the at least two cooperating nodes. The wireless device can adjust the receiver timing based on the composite received RS timing. A node specific RS can include a channel-state information reference signal (CSI-RS).

Abstract: Disclosed is a speaker diarization process for determining which speaker is speaking at what time during the course of a conversation. The entire process can be most easily described in five main parts: Segmentation where speech/non-speech decisions are made; frame feature extraction where useful information is obtained from the frames; segment modeling where the information from the frame feature extraction is combined with segment start and end time information to create segment specific features; speaker decisions when the segments are clustered to create speaker models; and corrections where frame level corrections are applied to the information extracted.

Abstract: An arrangement, for reproducing audio data of an acoustic scene, adapted for generating audio signals for at least a first and a second headphone channel of a headphone assembly, the audio signals corresponding to at least one audio object and/or sound source in the acoustic scene comprising at least one given close range and at least one given distant range arranged around a listener, the arrangement comprising a first headphone channel; a second headphone channel; a basic channel provider comprising at least a basic system adapted for reproducing audio signals corresponding to at least one audio object and/or sound source arranged in at least one distant range; a proximity channel provider comprising at least a proximity system adapted for reproducing audio signals corresponding to at least one audio object and/or sound source arranged in at least one close range.

Abstract: An audio signal processing device for downmixing of a first input signal and a second input signal to a downmix signal having: a dissimilarity extractor configured to receive the first input signal and the second input signal as well as to output an extracted signal, which is lesser correlated with respect to the first input signal than the second input signal and a combiner configured to combine the first input signal and the extracted signal in order to obtain the downmix signal.

Abstract: This application describes methods of signal processing and spatial audio synthesis. One such method includes accepting an auditory signal and generating an impression of auditory virtual reality by processing the auditory signal to impute a spatial characteristic on it via convolution with a plurality of head-related impulse responses. The processing is performed in a series of steps, the steps including: performing a first convolution of an auditory signal with a characteristic-independent, mixed-sign filter and performing a second convolution of the result of first convolution with a characteristic-dependent, sparse, non-negative filter. In some described methods, the first convolution can be pre-computed and the second convolution can be performed in real-time, thereby resulting in a reduction of computational complexity in said methods of signal processing and spatial audio synthesis.

Abstract: Apparatus and methods for reducing crosstalk in personal audio equipment are provided. In an example, a method to reduce headset audio crosstalk can include applying a first signal to a first speaker channel of a headset, coupling a second speaker channel to a first input of a comparator of a crosstalk compensation circuit using a first switch of the crosstalk compensation circuit, the switch and detect circuit including the crosstalk compensation circuit, coupling a first resistor divider to a second input of the comparator using a second switch of the crosstalk compensation circuit, and adjusting a resistance setting of the first resistor divider from an initial setting using an output of the comparator.

Abstract: A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.

Abstract: The audio amplifier includes a variable gain amplifier receiving the input audio signal and providing the output signal, whereby the output signal corresponds to the input signal amplified by a limiter gain. The audio amplifier further includes a limiter gain calculation unit, thus the input signal is amplified by the limiter gain. A control unit receives a signal representative of the input signal and is configured to estimate, based on a mathematical model, the input current or the total output current of the audio amplifier thus providing an estimated current signal corresponding to (and resulting from) the output signal, whereby the limiter gain calculation unit is configured to calculate, dependent on the estimation, the limiter gain such that the actual input current or the total output current of the audio amplifier does not exceed a threshold current value.

Abstract: The present technology relates to an acoustic signal processing device, an acoustic signal processing method, and a program capable of improving a feeling of localization of a sound image at a position deviated leftward or rightward from a median plane of a listener. A transaural processing unit performs a predetermined transaural process for an input signal by using a sound source opposite side HRTF and a sound source side HRTF to generate a first acoustic signal, and a second acoustic signal containing attenuated components in a first band which is the lowest band, and a second band which is the second lowest band in a range of a predetermined first frequency or higher frequencies, in bands of appearance of notches in the sound source opposite side HRTF. A subsidiary signal synthesis unit adds a first subsidiary signal constituted by a component in a predetermined band of the second acoustic signal to the first acoustic signal to generate a third acoustic signal.

Abstract: A method of interpolating a head-related transfer function (HRTF) and an audio output apparatus using the same are disclosed. The method includes receiving HRTF data corresponding to a point at which an altitude angle and an azimuth angle cross and receiving complementary information about a point at which the HRTF data is present, generating an HRTF interpolation signal corresponding to an altitude angle of a sound localization point, using HRTF data corresponding to two points constituting an altitude angle segment nearest the sound location point, calculating an amount of variation up to an azimuth angle ? of the sound localization point, using complementary information of two points constituting an azimuth angle segment nearest the sound localization point, and generating a final HRTF interpolation signal corresponding to the sound localization point by applying the amount of variation to the HRTF interpolation signal corresponding to the altitude angle of the sound localization point.

Abstract: The present disclosure is directed towards systems, methods and devices for tamper proofing documents by embedding data in a biometric identifier. The biometric identifier may be a real or synthetic fingerprint. In one non-limiting exemplary embodiment, a method for embedding data in a biometric identifier includes determining data to embed in the biometric identifier, determining at least one control parameter, and generating at least one biometric identifier feature. The method further includes including the at least one generated biometric identifier feature in the biometric identifier such that the determined data is embedded in the biometric feature. The determined data is based on a variable data stream. The control parameter controls the generation of at least one biometric identifier feature. The control parameter is based on the determined data. The generated biometric identifier feature is based on the determined control parameter.

Abstract: A hearing aid system includes: a first hearing aid comprising a first set of microphones, a first beamformer, a first processing module, and a first receiver; and a second hearing aid comprising a second set of microphones, a second beamformer, a second processing module, and a second receiver; wherein the first beamformer in a first operating mode is configured to provide a first audio signal in accordance with a first primary spatial characteristic, wherein the second beamformer in the first operating mode is configured to provide a second audio signal in accordance with a second primary spatial characteristic, the first primary spatial characteristic having a first main lobe with a first direction, and the second primary spatial characteristic having a second main lobe with a second direction, wherein the second direction is different from the first direction.

Abstract: An object detection and notification system includes an object detection system including logic that detects presence of a target object within a sensing range of a sensor and determines a direction of the target object relative to the autonomous vehicle. A notification system includes logic that provides a sound cue using a speaker to an occupant of the autonomous vehicle that is indicative of location of the target object relative to the autonomous vehicle.

Abstract: A recording method acquires, from a first musical instrument which outputs playing data which represents playing information, playing data of playing by the first musical instrument; generates a first waveform signal according to the played sounds of the first musical instrument which correspond to the playing data; generates a second waveform signal according to a sound including a sound emitted from the first musical instrument and other sounds; generates a third waveform signal wherein the first waveform signal is subtracted from the second waveform signal; generates audio data from the third waveform signal; and records the audio data.

Abstract: A directivity control method is provided for controlling a directivity of a sound collected by a first sound collector including a plurality of microphones. The directivity control method includes: forming a directivity of the sound in a direction toward a monitoring target corresponding to a first designated position in an image displayed on a display; obtaining information on a second designated position in the image displayed on the display, designated in accordance with a movement of the monitoring target, and changing the directivity of the sound toward the monitoring target corresponding to the second designated position by referring to the information on the second designated position.

Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by a new type of decode matrix for sound field data, and a new way to obtain the decode matrix.

Abstract: A sound reproduction system includes an electro-acoustic transducer and a transducer driver for driving the electro-acoustic transducer. The transducer drive includes a filter which is configured to reproduce at a listener's location an approximation to the local sound field that would be present at the listener's ears in recording space, taking into account the characteristics and intended position of the electro-acoustic transducer relative to the listener's ears. The electro-acoustic transducer includes a first sound emitter which provides an intermediate sound emission channel, and second and third sound emitters providing respective left and right sound emission channels. The first sound emitter is located intermediate of second and third sound emitters. Higher frequencies from at least one of the second and third sound emitters are transmitted closer to the first sound emitter while lower frequencies are transmitted away from the first sound emitter.

Abstract: A driving device for a sound system by loudspeaker signals, wherein the sound system has a wave field synthesis loudspeaker array and one or several supply loudspeakers arranged separate from the wave field synthesis array includes an audio input for receiving at least one audio signal from at least one sound source, a position input for receiving information on a position of the sound source, a wave field synthesis unit for calculating loudspeaker signals for the loudspeakers of the wave field synthesis loudspeaker array, and a provider for providing the loudspeaker signal for the one or the several supply loudspeakers. The driving device enables a sound system by means of which sound localization becomes possible for the audience and at the same time pleasant levels can be achieved also in the first rows of the audience.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: A method is provided for optimizing acoustic localization at one or more listening positions in a listening environment such as, but not limited to, a vehicle passenger compartment. The method includes generating a sound field with a group of loudspeakers assigned to at least one of the listening positions, the group of loudspeakers including first and second loudspeakers, where each loudspeaker is connected to a respective audio channel; calculating filter coefficients for a phase equalization filter; configuring a phase response for the phase equalization filter such that binaural phase difference (??mn) at the at least one of the listening positions or a mean binaural phase difference (m??mn) averaged over the listening positions is reduced in a predefined frequency range; and filtering the audio channel connected to the second loudspeaker with the phase equalization filter.

Abstract: Method, equipment and apparatus for acquiring a spatial audio direction vector, the method including: determining a position of a sound source in a multi-sound system; setting a parameter comprising: a human response time ?t and a tolerance percentage ?; acquiring a sound signal from the sound source; and processing the sound signal by using the parameter and acquiring a corresponding spatial audio direction vector {right arrow over (E)} within each time interval ?t. A proportional constant D is determined according to a modulus of a spatial audio direction vector {right arrow over (E)}, and provides spatial information of depth for a virtual image corresponding to a multi-tone audio signal. A vector angle ?E the spatial audio direction vector {right arrow over (E)} provides spatial information of direction for the virtual image corresponding to the multi-tone audio signal, to improve viewer's viewing experience.

Abstract: A method for compensating for acoustic crosstalk between a first and a second microphone unit being acoustically connected to a shared volume. The method includes the steps of providing a first output signal, Pout, from the first microphone unit, providing a second output signal, Uout, from the second microphone unit, and generating a compensated output signal by combining a portion of one of the output signals with the other output signal via addition or subtraction in order to compensate for acoustical crosstalk. The invention further relates to a microphone module configured to implement the before-mentioned method. The invention further relates to a hearing aid comprising the microphone module.

Abstract: A sound processing device includes a shift detection unit that detects a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and a correction unit that corrects a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point.

Abstract: Provided are methods and apparatus for enhancing a signal-to-noise ratio. In an example, provided is an apparatus configured to modify audio to better match the way the human brain processes audio by modifying the audio to a form which takes advantage of human echolocation capabilities. When humans listen to audio, they subconsciously listen for an echo and thus subconsciously focus on listening to, and for, meaningful information in audio. The focus causes humans to ignore noise in the audio, which results in enhancing a signal-to-noise ratio.

Abstract: A multichannel compensating audio system includes first and second compensation channels to psychoacoustically minimize deviations in a target response, to psychoacoustically move the physical position of a speaker and/or to psychoacoustically provide a substantially equal magnitude of sound from a plurality of speakers in a plurality of different listening positions. The first compensation channel may include a series connected delay circuit, a level adjuster circuit and a frequency equalizer circuit that generates a first compensated audio signal from a first audio signal. The second compensation channel may include a series connected delay circuit, a level adjuster circuit and a frequency equalizer circuit that generates a second compensated audio signal from a second audio signal. A first summing circuit is configured to receive at least the first audio signal and the second compensated audio signal and generate a first output signal for provision to a first speaker.

Abstract: Examples disclosed herein relate to an electronic gaming device including one or more processors, a memory, and one or more display devices. The electronic gaming device further includes a left surround audio device, a right surround audio device, a dialog enhancing center channel speaker, a low frequency effects device, a left speaker, and a right speaker. The combination of the left surround audio device; the right surround audio device; the dialog enhancing center channel speaker; the low frequency effects device; the left speaker; and/or the right speaker produce one or more sound effects in a vertical direction.

Abstract: Systems and methods for monitoring permission to be in a location comprising: a secured area having at least one entry point with a mechanical gate having an open and closed position; at least two wireless proximity sensors attached to a portion of (or the area adjacent to) the mechanical gate; a token device in communication with the wireless proximity sensors that determine a location of the token device relative to one of the wireless proximity sensors to provide a detection data point and a set of detection data points for the group of detection data points; and a system computing device that calculates the shared proximity of the token device. If the token device contains a valid ticket and the shared proximity of the token device is within a predetermined area the system computing device will cause the mechanical gate to go to the open position.

Abstract: A sound field measuring device (1) obtains frequency characteristics by collecting output sound outputted from a pair of speakers (101a, 101b) installed at a narrow interval. A low-pass filter (22a) extracts low-range components of a first measurement signal. A high-pass filter (22b) extracts mid/high-range components of a second measurement signal different from the first measurement signal. A combined signal generation unit (22c) generates a combined signal by combining the low-range components of the first measurement signal and the mid/high-range components of the second measurement signal. An external output unit (6) outputs the first measurement signal to an audio system (102). A microphone (7) collects the first measurement signal and the combined signal simultaneously outputted from the pair of speakers. A Fourier transform unit (13) obtains the frequency characteristics of a sound field by Fourier transforming the signals collected.