Abstract: A hearing device, e.g. a hearing aid, comprises a) first and b) second input transducers located on the head, and at or in an ear canal of the user, respectively, and providing respective electric input signals, c) a signal processing unit comprising c1) a weighting unit for applying weights to the electric input signals, and c2) a hearing loss processing unit providing a processed signal. The hearing device further comprises d) an output unit to provide a stimulus perceivable by the user as sound, e) a feedback detection unit for providing a measure of the current level of feedback from the output to the input, and f) an input signal weight control unit configured to control or influence first and second weights applied to the first and second electric input signals in dependence of the measure of the current level of feedback, and a current level and frequency dependent target gain.

Abstract: An audio circuit with an integral hearing test is disclosed. The circuit includes at least one variable gain amplifier (VGA) coupled to receive an audio signal and a plurality of filters. Each filter is coupled to the at least one VGA and configured to filter an output signal from the at least one VGA. A processor is coupled to the VGAs and configured to apply a selected frequency to the at least one VGA in a test mode and to control a gain of the at least one VGA in a normal mode.

Abstract: Described herein are methods and devices for mixing input signals from multiple audio input sources in a hearing device such as a hearing aid. In one embodiment, a gain is applied to a telecoil signal where the gain is computed so as to depend on the level of a microphone signal. The technique does not depend on the user's audiogram and allows control of the gain for all environments with one adjustment.

Abstract: A hearing device, e.g.

Abstract: The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system configured to generate a bitstream indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system comprises a downmix processing unit configured to generate the downmix signal from a multi-channel input signal; wherein the downmix signal comprises m channels and wherein the multi-channel input signal comprises n channels; n, m being integers with m<n. Furthermore, the system comprises a parameter processing unit configured to determine the spatial metadata from the multi-channel input signal.

Abstract: A hearing device comprises A) a forward path, comprising a1) an input unit for providing a time-domain electric input signal as digital samples, a2) an analysis filter bank configured to provide a time-frequency representation of said electric input signal, a3) a signal processing unit for processing a signal of the forward path and providing a number of processed channel-signals, B) an onset detector configured to receive said time-domain electric input signal before entering said analysis filter bank, and to provide an onset control signal dependent on a current first order derivative of an envelope thereof, C) a level estimation unit for estimating a current level of said frequency sub-band signals, and comprising c1) a level adjustment unit configured to adjust the current levels of said frequency sub-band signals, and to control said level adjustment in dependence of said onset control signal. The invention may be used in audio devices, e.g. hearing aids.

Abstract: The signal processing device comprises the adaptive gain calculation unit configured to calculate and output the adaptive gain for processing the voice reception output signal, by utilizing at least the voice transmission input signal and the relative volume value indicating the predetermined relative sound volume ratio between the voice transmission input signal and the voice reception output signal; and the multiplication unit configured to calculate the voice reception output signal by multiplying the signal based on the voice reception input signal by the value based on the adaptive gain.

Abstract: An audio control circuit is provided in the present disclosure. The audio control circuit includes an input module, a volume signal detecting module and a volume adjusting module. The input module is used for receiving an audio signal. The volume signal detecting module is used for receiving a volume control signal and generates a volume adjusting signal according to the volume control signal. The volume adjusting module is used for receiving the audio signal and the volume adjusting signals and generates an amplified audio signal according to the audio signal and the volume adjusting signal. The volume control signal is a pulse width modulation signal.

Abstract: An audio enhancement system can provide spatial enhancement, low frequency enhancement, and/or high frequency enhancement for headphone audio. The spatial enhancement can increase the sense of spaciousness or stereo separation between left and right headphone channels. The low frequency enhancement can enhance bass frequencies that are unreproducible or attenuated in headphone speakers by emphasizing harmonics of the low bass frequencies. The high frequency enhancement can emphasize higher frequencies that may be less reproducible or poorly tuned for headphone speakers. In some implementations, the audio enhancement system provides a user interface that enables a user to control the amount (e.g., gains) of each enhancement applied to headphone input signals. The audio enhancement system may also be designed to provide one or more of these enhancements more effectively when headphones with good coupling to the ear are used.

Abstract: A portable full duplex intercom system using Bluetooth and method of use are provided in which the intercom system has a built in microphone and speaker in a device. In one embodiment, the portable full duplex intercom system using Bluetooth may be used in a vehicle to allow the people in the vehicle to communicate with each other.

Abstract: A method for dynamically adjusting recovery time in wide dynamic range compression (WDRC) for use in a hearing aid is provided. The method includes the steps of: receiving an input signal via the hearing aid; applying a band-pass filter on the input acoustic signal to calculate a high-energy ratio; calculating an over-zero rate ratio corresponding to the input acoustic signal; calculating a consonant occurring probability according to the high-frequency energy ratio and the over-zero rate ratio; applying a consonant determination mechanism on the input acoustic signal, and adjusting the consonant occurring probability according to the results of the consonant determination mechanism; calculating a recovery time factor corresponding to the input acoustic signal according to the adjusted consonant occurring probability; and performing a WDRC process on the input acoustic signal according to the recovery time factor to generate an output acoustic signal.

Abstract: A digital microphone is provided. The digital microphone includes an acoustic sensor, a bias generator, first and second attenuators, a buffer, an amplifier, an ADC, and a controller. The acoustic sensor transfers an acoustic signal to a voltage signal. The bias generator provides a bias voltage to the acoustic sensor. The first attenuator attenuates the voltage signal by a first attenuation value. The buffer buffers the voltage signal to generate a buffered voltage signal. The amplifier amplifies the buffered voltage signal to generate an amplified signal. The ADC converts the amplified signal to a data signal with a digital format. The second attenuator attenuates the data signal by a second attenuation value. The controller determines whether the amplified signal is larger than a reference value and adjusts the bias voltage and the first and second attenuation values of the first and second attenuators according to the result determined by the controller.

Abstract: A hearing loss compensation apparatus and method compensate for distortion caused by a change in a sound transmission path using hearing characteristics of a user. The hearing loss compensation apparatus may include a sound direction detection device configured to detect a sound generation direction, in which a sound generates, using one or more microphones, and a sound compensation device configured to compensate the sound using hearing characteristics of a user corresponding to the sound generation direction. Additionally, a hearing characteristics measurement apparatus and method provide a way to obtain such hearing characteristics information.

Abstract: A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of medical device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

Abstract: Useful and meaningful machine characteristic information may be derived through analysis of oversampled digital data collected using dynamic signal analyzers, such as vibration analyzers. Such data have generally been discarded in prior art systems. In addition to peak values and decimated values, other oversampled values are used that are associated with characteristics of the machine being monitored and the sensors and circuits that gather the data. This provides more useful information than has previously been derived from oversampled data within a sampling interval.

Abstract: An electronic apparatus and a method for controlling the electronic apparatus are provided. The electronic apparatus includes an audio processor that outputs an audio including a first audio component and a second audio component and a controller that controls the audio processor to change a ratio of the first audio component to the second audio component according to the changed audio level, in response to receiving an instruction to change a level of the output audio.

Abstract: A system is provided that dynamically converts an input signal into a haptic signal. The system generates effect objects, where an effect object includes an instruction to perform a haptic conversion algorithm on the input signal to convert the input signal into an output signal, and where an order of the effect objects is defined. The system further receives the input signal. The system further applies the effect objects to the input signal in the defined order, where the output signal of an effect object forms the haptic signal. The system further sends the haptic signal to a haptic output device, where the haptic signal causes the haptic output device to output haptic effects.

Abstract: An apparatus for mapping a first input channel and a second input channel of an input channel configuration to at least one output channel of an output channel configuration, wherein each input channel and each output channel has a direction in which an associated loudspeaker is located relative to a central listener position, configured to map the first input channel to a first output channel of the output channel configuration; and despite of the fact that an angle deviation between a direction of the second input channel and a direction of the first output channel is less than an angle deviation between a direction of the second input channel and the second output channel and/or is less than an angle deviation between the direction of the second input channel and the direction of the third output channel, map the second input channel to the second and third output channels by panning between the second and third output channels.

Abstract: The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. The encoding device stores speaker arrangement information in a comment region in a PCE of an encoded bit stream and stores a synchronous word and identification information in the comment region such that other public comments and the speaker arrangement information stored in the comment region can be distinguished from each other. When an encoded bit stream is decoded, it is determined whether the speaker arrangement information is stored on the basis of the synchronous word and the identification information stored in the comment region. Audio data included in the encoded bit stream is output according to the arrangement of the speakers corresponding to the determination result. The present technique can be applied to an encoding device.

Abstract: A system and method for automatically controlling the timbre of a sound signal in a listening room are also disclosed, which include the following: generating an acoustic sound output from an electrical sound signal; measuring the total acoustic sound level in the room and generating an electrical total sound signal representative of the total acoustic sound level in the room, wherein the total acoustic sound comprises the acoustic sound output generated from the electrical sound signal; and adjusting the gain of the electrical sound signal dependent on a room-dependent gain signal, the room-dependent gain signal being determined from reference room data and estimated room data.

Abstract: A hearing aid device that is configured to be inserted into the bony region of an ear canal includes a receiver, a microphone, a processor for processing sound signals detected by the microphone, and a feedback suppressor for carrying out electronic feedback suppression by applying at least one of a frequency shift and a time delay to the sound signals detected by the microphone. The hearing aid device may be configured to carry out the electronic feedback suppression including at least one of the frequency shift and the time delay as function of the frequency of the detected sound signals.

Abstract: A processing method includes acquiring an audio signal, magnifying one or more first portions of the audio signal in response to one or more magnitudes of the one or more first portions of the audio signal are lower than a predetermined threshold, deforming one or more second portions of the audio signal in response to one or more magnitudes of the one or more second portions of the audio signal are greater than the predetermined threshold, and performing a noise reduction to the one or more deformed second portions of the audio signal to reduce the one or more deformed second portions of the audio signal.

Abstract: A meter includes a hermetically encapsulated electronic metering mechanism having a meter unit for the determination of meter readings, the metering mechanism including a data memory for storing the meter readings, and the metering unit including an antenna of a defined shape, and a readout unit arranged outside the metering mechanism for reading the meter readings from the data memory. The meter is operated according to a method for the determination of meter readings and for the wireless transmission of electrical energy. The shape of the readout unit antenna is identical to the shape of the metering unit antenna, wherein the congruent and predefined positioning of the readout unit antenna ensures an effective wireless energy supply to the metering unit by electromagnetic radiation through the readout unit, and, independently thereof, a stable wireless data coupling for the determination of meter readings between the metering unit and the readout unit.

Abstract: A hearing aid (21) includes a signal processor producing signals that have been processed to compensate for a hearing impairment, a first output converter (26), a second output converter (27), a first acoustic output transducer (34) and at least a second output transducer (35). The first output converter (26) and the first output transducer (34) are configured to reproduce the high frequencies of the processed signals, and the second output converter (27) and the second output transducer (35) are configured to reproduce the low frequencies of the processed signals. The output converters (26, 27) may preferably be embodied as direct digital drive output converters. The processed signals are split between the first and second output converters according to a cross-over frequency tunable by programming.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. A filtering loudness gain of a filter signal of each frequency band is adjusted according to a wide dynamic range compression curve without an upper output loudness limit. The filtering loudness gain of each frequency band is reduced by lowering a gain decrease adjustment value, so as to reduce a loudness of a loudness adjusted filter signal, and thus a loudness of an output voice signal is lower than a first threshold value.

Abstract: Multi-channel audio content is mixed for a particular loudspeaker setup. However, a consumer's audio setup is very likely to use a different placement of speakers. The present invention provides a method of rendering multi-channel audio that assures replay of the spatial signal components with equal loudness of the signal. A method for obtaining an energy preserving mixing matrix (G) for mixing L1 input audio channels to L2 output channels comprises steps of obtaining a first mixing matrix ?, performing a singular value decomposition on the first mixing matrix ? to obtain a singularity matrix S, processing the singularity matrix S to obtain a processed singularity matrix ?, determining a scaling factor a, and calculating an improved mixing matrix G according to G=aU?VT. The perceived sound, loudness, timbre and spatial impression of multi-channel audio replayed on an arbitrary loudspeaker setup practically equals that of the original speaker setup.

Abstract: A circuit can be used in a speaker system. The circuit includes an amplifier with an output configured to be coupled to a speaker. An offset comparator has an input coupled the output of the amplifier and is configured to provide an offset control signal. A digital circuit has a first input coupled to an output of the offset comparator, a second input configured to receive an amplifier control signal, a third input configured to receive a play control signal, and an output configured to provide a forced mute signal that can be used to control the amplifier.

Abstract: A digital compressor for compressing an input audio signal is presented. The digital compressor comprises a compression gain control for providing a compression gain parameter, and a compression parameter determiner for determining a compression ratio from the compression gain parameter. The compression parameter determiner may be configured to weight the compression gain parameter by a predetermined weight factor to obtain the compression ratio. The digital compressor further comprises an auxiliary signal generator for manipulating the input audio signal in dependence of the compression ratio to obtain a first auxiliary signal, and a combiner unit for combining the first auxiliary signal with the compression gain parameter to obtain a second auxiliary signal, and for combining the input audio signal with the second auxiliary signal to obtain the compressed audio signal.

Abstract: Methods and apparatuses for user sound exposure limiting are disclosed. In one example, an accumulated sound dose exposure from a headset speaker is determined for a plurality of sequentially monitored time intervals during a current listening session, wherein the current listening session comprises a total session time. It is determined whether a predicted sound dose exposure for the total session time exceeds or falls below a permitted sound dose exposure limit, the predicted sound dose exposure determined from the accumulated sound dose exposure. A threshold intervention level at which a time-weighted-average limiter at a headset applies attenuation to an audio signal output at a headset speaker is adjusted.

Abstract: The present invention provides control systems and methods for a power inverter. For example, a control system comprises a plurality of sensors and a controller. The plurality of sensors are configured to measure electrical signals that are indicative of output voltages and output currents of the power inverter. The controller, coupled to the power inverter, is configured to: determine a target power based on real power frequency droop information and a first frequency if the power inverter is in a voltage source mode; determine a target power based on a power limit and a predetermined power command if the power inverter is in a current source mode; and generate a second frequency based on the target power, a measured power, and a latency estimate of a simulated generator. The second frequency is used to control the output power of the power inverter.

Abstract: A detection circuit (1) comprises a first and a second contact terminal (P1, P2) for connecting a microphone (HM) with a defined polarity. Furthermore, a first and a second switch (S1, S2) are provided and respectively connect the first and the second contact terminal (P1, P2) to a reference potential terminal (GND). A supply terminal (SUP) for supplying an input signal is either connected to the first contact terminal (P1) or to the second contact terminal (P2) via a third switch (S3). A measuring device (MS) for acquiring a measurement signal in response to the supplied input signal is coupled to a connection between the supply terminal (SUP) and the third switch (S3).

Abstract: A method for operating a hearing device for a user having tinnitus includes converting an acoustic input signal into a corresponding converted input signal, applying a transfer function to the converted input signal for generating a processed input signal, generating a stimulation signal depending on characteristics of a perceived disturbing internal signal resulting from a tinnitus perceived by a prospective user of the hearing device, limiting the stimulation signal by applying a predefined limiting scheme thereby generating a limited stimulation signal, superimposing the limited stimulation signal onto the processed signal thereby generating an output signal, and feeding the output signal to an output transducer for generating a hearing device output signal.

Abstract: An audio system includes a processor including an input configured to: receive a baseband audio signal and modulate the baseband audio signal to create a modulated audio signal comprising audio signal frequency components in a first frequency range; clip the modulated audio signal to create a clipped, modulated audio signal the clipped modulated audio signal comprising the audio signal frequency components in the first range and further comprising distortion frequency components outside the first frequency range. The system can further be configured to filter the clipped, modulated audio signal to remove frequency components outside the first frequency to remove distortion components outside that frequency range.

Abstract: A filter gain compensation method for a specific frequency band for use in an electronic device is provided. In the method, a windowed filter is applied on each of a plurality of band-pass filters in a multi-segment band-pass filter corresponding to the high-frequency signal and different frequency bands of the low-frequency signal. In addition, a high-frequency cancellation filter corresponding to the high-frequency signal of an input digital signal is calculated. A compensation gain for the high-frequency cancellation filter and each of the band-pass filters is calculated according to fitting frequency gains and a fitting frequency relationship matrix. The output acoustic signal is synthesized using output signals from the windowed band-pass filters and the high-frequency cancellation filter that are obtained according to updated filter features and compensation gain for the windowed band-pass filters and the high-frequency cancellation filter.

Abstract: A speech synthesis apparatus includes a content selection unit that selects a text content item to be converted into speech; a related information selection unit that selects related information which can be at least converted into text and which is related to the text content item selected by the content selection unit; a data addition unit that converts the related information selected by the related information selection unit into text and adds text data of the text to text data of the text content item selected by the content selection unit; a text-to-speech conversion unit that converts the text data supplied from the data addition unit into a speech signal; and a speech output unit that outputs the speech signal supplied from the text-to-speech conversion unit.

Abstract: A method of processing telephone sound is disclosed. Before the telephone sound is transmitted, the frequencies of high frequency consonants are lowered. If the recipient has severe hearing impairment, the frequencies of the high frequency consonants are lowered again.

Abstract: A headset includes a wearable body, first and second earphones extending from the wearable body, controls for controlling an external communication/multimedia device wirelessly, a microphone for picking up vocal data from a user of the headset system and a signal processing unit. The signal processing unit includes circuitry for processing the vocal data into a distinctly audible vocal feedback signal, circuitry for enhancing the vocal feedback signal thereby producing an enhanced vocal feedback signal and circuitry for mixing the enhanced vocal feedback signal with audio signals originating from the external communication/multimedia device, thereby producing a mixed output signal and then sending the mixed output signal to the user via the earphones. The external communication/multimedia device comprises a vocal command application and the headset further comprises a vocal command control for sending vocal commands to the external communication/multimedia device and to the vocal command application.

Abstract: A computer-implemented method comprising: determining one or more features of a subject signal; revising one or more control signals on the basis of the one or more features; modifying a level of the subject signals based on the control signals. At least one of the features is determined by: comparing a given one of the subject signals against a boundary signal to produce a corresponding given boundary comparison signal; and summarizing the behavior of the given boundary comparison signal over a time interval.

Abstract: An improved automatic loudness control system and method comprise controlling gain/attenuation applied to an input audio signal and providing an output audio signal that is the amplified/attenuated input audio signal; evaluating an actual loudness of the input audio signal from the input audio signal and a desired loudness of the input audio signal from a volume control input; and evaluating the gain/attenuation applied to the input audio signal from the actual loudness and the desired loudness of the input audio signal.

Abstract: A hearing aid includes: an input transducer for generating an audio signal; a feedback suppression circuit configured for modelling a feedback path of the hearing aid; a subtractor for subtracting an output signal of the feedback suppression circuit from the audio signal to form a feedback compensated audio signal; a signal processor that is coupled to an output of the subtractor for processing the feedback compensated audio signal to perform hearing loss compensation; and a receiver that is coupled to an output of the signal processor for converting the processed feedback compensated audio signal into a sound signal; wherein the hearing aid further comprises a gain processor for performing gain adjustment of the feedback compensated audio signal based at least on an estimate of a residual feedback signal of the feedback compensated audio signal, wherein the estimate of the residual feedback signal is based at least on the audio signal.

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: A dynamic range compressor includes an input terminal for receiving input signal to be compressed, an amplifier unit for amplifying the signal to be compressed by an amplification factor, for deriving a compressed output signal, an output terminal for supplying the compressed output signal, a first envelope detector unit for deriving a first envelope signal from the input signal, and an amplifier control unit for generating an amplifier control signal in dependence of an envelope signal. Further including a second envelope detector unit for deriving a second envelope signal from the input signal, a first signal level prediction unit for generating a first prediction signal from the first envelope signal, a second signal level prediction unit for generating a second prediction signal from the second envelope signal, and a signal combination unit for combining the first and second prediction signals to generate a combined prediction signal.

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: A method of audio signal processing includes receiving a first audio input signal (first input signal) at an input of a first integrating amplifier of a first single-ended (SE) closed loop channel, and second input signal with a polarity reversed relative to the first input signal at an input of a second integrating amplifier configured of a second SE closed loop channel. During audio signal processing a common-mode (CM) reference voltage level applied to a current source coupled to an input of the first and second integrating amplifiers is dynamically changed including whenever a level of the input signals is below a predetermined low level, reducing the CM reference voltage level for implementing low duty cycle (LDC) PWM operation, and whenever the level is above a level that corresponds to an onset of clipping, increasing the CM reference voltage level for at least reducing the clipping to lower crossover distortion.

Abstract: A loudness limiting method for attenuating an input audio signal in order to constrain a resulting loudness level of an output audio signal by a pre-determined threshold loudness level, the method involving determining a time-varying loudness level of the input audio signal by means of a finite-length sliding window, calculating a time-varying attenuation gain in accordance with a pre-determined threshold loudness level, time-delaying the input audio signal, and applying the time-varying attenuation gain to the time-delayed input audio signal resulting in the output audio signal. A loudness signal processor carries out the loudness limiting method and a computer program product enables a computer to carry out the loudness limiting method.

Abstract: Systems, devices, and methods are provided to inhibit apparent amplitude modulation in non-linear processing that causes distortion in a processed signal. One aspect of the invention includes a hearing aid. The hearing aid includes a microphone to receive an input signal, a speaker to reproduce the input signal, and a processor. The processor processes the input signal using a gain. The processor includes an inhibitor, which inhibits distortions, and an adjuster, which adjusts the gain. The inhibitor acts to smooth an envelope of the input signal to inhibit undesired modulation. The adjuster adjusts the gain if the envelope is either above or below a threshold.

Abstract: A novel audio ducking method that is aware of the loudness levels of the audio content is provided. The method specifies a minimum loudness separation between audio tracks that are designated as masters and audio tracks that are designated as slaves. The method attenuates the volume of the slave tracks in order to provide at least the minimum loudness separation between the slave tracks and the master tracks. The amount of attenuation for a slave is determined based on the loudness levels of the slave and of a master.

Abstract: The disclosed apparatus extracts particular instrument sounds included in music in real time during music reproduction. The audio processing section 12 has an attack decision section 42 that decides attack positions of the bass drum sounds in audio signals, a bass drum sounds decision section 43 that decides sound periods of the bass drum in the audio signals, and a bass drum sounds extraction section 44 that extracts bass drum sounds using a time varying type filter for the attack positions and an adaptive type filter for the sound periods.

Abstract: A multi-band audio compressor that may provide not only better and brighter sound, but also speaker protection. The multi-band audio compressor breaks an input audio signal into different frequency bands. For each band signal, a volume re-mapper translates a user preference volume level to a converted volume level based on a programmable volume curve for the band signal. For each frequency band, the band signal is processed by a gain stage and a compressor. Each gain stage applies a signal gain to the band signal based on the converted volume level. Each compressor compresses the output of the gain stage. After compression, the different frequency band signals are re-combined and the combined audio signal may then be passed to a power amplifier that is driving a speaker. Other embodiments are also described and claimed.

Abstract: A sound separation device includes: a signal obtainment unit which obtains a plurality of acoustic signals including a first acoustic signal and a second acoustic signal; a differential signal generation unit which generates a differential signal that is a signal representing a difference in a time domain between the first acoustic signal and the second acoustic signal; an acoustic signal generation unit which generates, using at least one acoustic signal among the acoustic signals, a third acoustic signal; and an extraction unit which generates a frequency signal by subtracting, from a signal obtained by transforming the third acoustic signal into a frequency domain, a signal obtained by transforming the differential signal into a frequency domain, and generates a separated acoustic signal by transforming the generated frequency signal into a time domain.